[Jul 6 15:51:30] VERBOSE[2904] config.c: == Parsing '/etc/asterisk/logger.conf': [Jul 6 15:51:30] DEBUG[2904] config.c: Parsing /etc/asterisk/logger.conf [Jul 6 15:51:30] VERBOSE[2904] config.c: == Found [Jul 6 15:51:30] VERBOSE[2904] logger.c: Asterisk Event Logger restarted [Jul 6 15:51:30] VERBOSE[2904] logger.c: Asterisk Queue Logger restarted [Jul 6 15:51:35] DEBUG[2782] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Jul 6 15:51:35] DEBUG[2782] chan_sip.c: Trying to put 'SIP/2.0 401' onto UDP socket destined for 10.0.5.198:5060 [Jul 6 15:51:35] DEBUG[2782] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Jul 6 15:51:35] DEBUG[2782] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.0.5.198:5060 [Jul 6 15:51:35] DEBUG[2689] devicestate.c: No provider found, checking channel drivers for SIP - 0004f215aabb-1 [Jul 6 15:51:35] DEBUG[2689] chan_sip.c: Checking device state for peer 0004f215aabb-1 [Jul 6 15:51:35] DEBUG[2689] devicestate.c: Changing state for SIP/0004f215aabb-1 - state 1 (Not in use) [Jul 6 15:51:35] DEBUG[2689] devicestate.c: device 'SIP/0004f215aabb-1' state '1' [Jul 6 15:51:35] DEBUG[2779] app_queue.c: Device 'SIP/0004f215aabb-1' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Jul 6 15:51:36] DEBUG[2782] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Jul 6 15:51:36] DEBUG[2782] chan_sip.c: Trying to put 'SIP/2.0 401' onto UDP socket destined for 10.0.5.198:5060 [Jul 6 15:51:36] DEBUG[2782] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Jul 6 15:51:36] DEBUG[2782] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.0.5.198:5060 [Jul 6 15:51:36] DEBUG[2689] devicestate.c: No provider found, checking channel drivers for SIP - 0004f215aabb-2 [Jul 6 15:51:36] DEBUG[2689] chan_sip.c: Checking device state for peer 0004f215aabb-2 [Jul 6 15:51:36] DEBUG[2689] devicestate.c: Changing state for SIP/0004f215aabb-2 - state 1 (Not in use) [Jul 6 15:51:36] DEBUG[2689] devicestate.c: device 'SIP/0004f215aabb-2' state '1' [Jul 6 15:51:36] DEBUG[2779] app_queue.c: Device 'SIP/0004f215aabb-2' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Jul 6 15:51:52] DEBUG[2782] chan_sip.c: Auto destroying SIP dialog 'c8ce19c6966b23d9' [Jul 6 15:51:52] DEBUG[2782] chan_sip.c: Destroying SIP dialog c8ce19c6966b23d9 [Jul 6 15:51:57] DEBUG[2782] acl.c: Found IP address for this socket [Jul 6 15:51:57] DEBUG[2782] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 10.0.5.191:5060 [Jul 6 15:51:57] DEBUG[2782] chan_sip.c: Allocating new SIP dialog for 47874955-7183601f-83d054e4@10.0.5.198 - SUBSCRIBE (No RTP) [Jul 6 15:51:57] DEBUG[2782] chan_sip.c: **** Received SUBSCRIBE (10) - Command in SIP SUBSCRIBE [Jul 6 15:51:57] DEBUG[2782] chan_sip.c: build_route: Contact hop: [Jul 6 15:51:57] DEBUG[2782] chan_sip.c: Trying to put 'SIP/2.0 401' onto UDP socket destined for 10.0.5.198:5060 [Jul 6 15:51:57] DEBUG[2782] chan_sip.c: **** Received SUBSCRIBE (10) - Command in SIP SUBSCRIBE [Jul 6 15:51:57] DEBUG[2782] chan_sip.c: build_route: Retaining previous route: [Jul 6 15:51:57] DEBUG[2782] chan_sip.c: Trying to put 'SIP/2.0 404' onto UDP socket destined for 10.0.5.198:5060 [Jul 6 15:51:57] DEBUG[2782] chan_sip.c: Destroying SIP dialog 47874955-7183601f-83d054e4@10.0.5.198 [Jul 6 15:52:05] VERBOSE[2782] chan_sip.c: <--- SIP read from UDP:10.0.5.196:5060 ---> REGISTER sip:10.0.5.191:5060 SIP/2.0 Via: SIP/2.0/UDP 10.0.5.196:5060;branch=z9hG4bK8a837684eca86a2b0.efc6da06ebc63beb9 Max-Forwards: 70 From: ;tag=d191d653b3 To: Call-ID: c8ce19c6966b23d9 CSeq: 26480 REGISTER Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Authorization: Digest username="33i",realm="asterisk",nonce="23ca8b56",uri="sip:10.0.5.191:5060",response="cfca5a08addaa5243d3e7118b84982a5",algorithm=MD5 Contact: "33i" ;expires=60;+sip.instance="" Supported: gruu, path User-Agent: Aastra 9143i/2.5.3.2002 Content-Length: 0 <-------------> [Jul 6 15:52:05] DEBUG[2782] chan_sip.c: Header 0 [ 36]: REGISTER sip:10.0.5.191:5060 SIP/2.0 [Jul 6 15:52:05] DEBUG[2782] chan_sip.c: Header 1 [ 82]: Via: SIP/2.0/UDP 10.0.5.196:5060;branch=z9hG4bK8a837684eca86a2b0.efc6da06ebc63beb9 [Jul 6 15:52:05] DEBUG[2782] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jul 6 15:52:05] DEBUG[2782] chan_sip.c: Header 3 [ 46]: From: ;tag=d191d653b3 [Jul 6 15:52:05] DEBUG[2782] chan_sip.c: Header 4 [ 29]: To: [Jul 6 15:52:05] DEBUG[2782] chan_sip.c: Header 5 [ 25]: Call-ID: c8ce19c6966b23d9 [Jul 6 15:52:05] DEBUG[2782] chan_sip.c: Header 6 [ 20]: CSeq: 26480 REGISTER [Jul 6 15:52:05] DEBUG[2782] chan_sip.c: Header 7 [ 88]: Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO [Jul 6 15:52:05] DEBUG[2782] chan_sip.c: Header 8 [ 53]: Allow-Events: talk, hold, conference, LocalModeStatus [Jul 6 15:52:05] DEBUG[2782] chan_sip.c: Header 9 [154]: Authorization: Digest username="33i",realm="asterisk",nonce="23ca8b56",uri="sip:10.0.5.191:5060",response="cfca5a08addaa5243d3e7118b84982a5",algorithm=MD5 [Jul 6 15:52:05] DEBUG[2782] chan_sip.c: Header 10 [129]: Contact: "33i" ;expires=60;+sip.instance="" [Jul 6 15:52:05] DEBUG[2782] chan_sip.c: Header 11 [ 21]: Supported: gruu, path [Jul 6 15:52:05] DEBUG[2782] chan_sip.c: Header 12 [ 35]: User-Agent: Aastra 9143i/2.5.3.2002 [Jul 6 15:52:05] DEBUG[2782] chan_sip.c: Header 13 [ 17]: Content-Length: 0 [Jul 6 15:52:05] DEBUG[2782] chan_sip.c: Header 14 [ 0]: [Jul 6 15:52:05] VERBOSE[2782] chan_sip.c: --- (14 headers 0 lines) --- [Jul 6 15:52:05] DEBUG[2782] acl.c: Found IP address for this socket [Jul 6 15:52:05] DEBUG[2782] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 10.0.5.191:5060 [Jul 6 15:52:05] DEBUG[2782] chan_sip.c: Allocating new SIP dialog for c8ce19c6966b23d9 - REGISTER (No RTP) [Jul 6 15:52:05] DEBUG[2782] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Jul 6 15:52:05] DEBUG[2782] chan_sip.c: Initializing initreq for method REGISTER - callid c8ce19c6966b23d9 [Jul 6 15:52:05] VERBOSE[2782] chan_sip.c: Sending to 10.0.5.196 : 5060 (no NAT) [Jul 6 15:52:05] VERBOSE[2782] chan_sip.c: <--- Transmitting (no NAT) to 10.0.5.196:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.0.5.196:5060;branch=z9hG4bK8a837684eca86a2b0.efc6da06ebc63beb9;received=10.0.5.196 From: ;tag=d191d653b3 To: ;tag=as4d33ffaa Call-ID: c8ce19c6966b23d9 CSeq: 26480 REGISTER Server: Asterisk PBX 1.6.2.9 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="251d90bf" Content-Length: 0 <------------> [Jul 6 15:52:05] DEBUG[2782] chan_sip.c: Trying to put 'SIP/2.0 401' onto UDP socket destined for 10.0.5.196:5060 [Jul 6 15:52:05] VERBOSE[2782] chan_sip.c: Scheduling destruction of SIP dialog 'c8ce19c6966b23d9' in 32000 ms (Method: REGISTER) [Jul 6 15:52:05] VERBOSE[2782] chan_sip.c: <--- SIP read from UDP:10.0.5.196:5060 ---> REGISTER sip:10.0.5.191:5060 SIP/2.0 Via: SIP/2.0/UDP 10.0.5.196:5060;branch=z9hG4bK0cbb7cc3ed9391b3d.39503faaa978ade2f Max-Forwards: 70 From: ;tag=d191d653b3 To: Call-ID: c8ce19c6966b23d9 CSeq: 26481 REGISTER Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Authorization: Digest username="33i",realm="asterisk",nonce="251d90bf",uri="sip:10.0.5.191:5060",response="0069b554a4f240f71e4c115d67ef231e",algorithm=MD5 Contact: "33i" ;expires=60;+sip.instance="" Supported: gruu, path User-Agent: Aastra 9143i/2.5.3.2002 Content-Length: 0 <-------------> [Jul 6 15:52:05] DEBUG[2782] chan_sip.c: Header 0 [ 36]: REGISTER sip:10.0.5.191:5060 SIP/2.0 [Jul 6 15:52:05] DEBUG[2782] chan_sip.c: Header 1 [ 82]: Via: SIP/2.0/UDP 10.0.5.196:5060;branch=z9hG4bK0cbb7cc3ed9391b3d.39503faaa978ade2f [Jul 6 15:52:05] DEBUG[2782] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jul 6 15:52:05] DEBUG[2782] chan_sip.c: Header 3 [ 46]: From: ;tag=d191d653b3 [Jul 6 15:52:05] DEBUG[2782] chan_sip.c: Header 4 [ 29]: To: [Jul 6 15:52:05] DEBUG[2782] chan_sip.c: Header 5 [ 25]: Call-ID: c8ce19c6966b23d9 [Jul 6 15:52:05] DEBUG[2782] chan_sip.c: Header 6 [ 20]: CSeq: 26481 REGISTER [Jul 6 15:52:05] DEBUG[2782] chan_sip.c: Header 7 [ 88]: Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO [Jul 6 15:52:05] DEBUG[2782] chan_sip.c: Header 8 [ 53]: Allow-Events: talk, hold, conference, LocalModeStatus [Jul 6 15:52:05] DEBUG[2782] chan_sip.c: Header 9 [154]: Authorization: Digest username="33i",realm="asterisk",nonce="251d90bf",uri="sip:10.0.5.191:5060",response="0069b554a4f240f71e4c115d67ef231e",algorithm=MD5 [Jul 6 15:52:05] DEBUG[2782] chan_sip.c: Header 10 [129]: Contact: "33i" ;expires=60;+sip.instance="" [Jul 6 15:52:05] DEBUG[2782] chan_sip.c: Header 11 [ 21]: Supported: gruu, path [Jul 6 15:52:05] DEBUG[2782] chan_sip.c: Header 12 [ 35]: User-Agent: Aastra 9143i/2.5.3.2002 [Jul 6 15:52:05] DEBUG[2782] chan_sip.c: Header 13 [ 17]: Content-Length: 0 [Jul 6 15:52:05] DEBUG[2782] chan_sip.c: Header 14 [ 0]: [Jul 6 15:52:05] VERBOSE[2782] chan_sip.c: --- (14 headers 0 lines) --- [Jul 6 15:52:05] DEBUG[2782] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Jul 6 15:52:05] DEBUG[2782] chan_sip.c: Initializing initreq for method REGISTER - callid c8ce19c6966b23d9 [Jul 6 15:52:05] VERBOSE[2782] chan_sip.c: Sending to 10.0.5.196 : 5060 (no NAT) [Jul 6 15:52:05] VERBOSE[2782] chan_sip.c: <--- Transmitting (no NAT) to 10.0.5.196:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.5.196:5060;branch=z9hG4bK0cbb7cc3ed9391b3d.39503faaa978ade2f;received=10.0.5.196 From: ;tag=d191d653b3 To: ;tag=as4d33ffaa Call-ID: c8ce19c6966b23d9 CSeq: 26481 REGISTER Server: Asterisk PBX 1.6.2.9 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Expires: 60 Contact: ;expires=60 Date: Tue, 06 Jul 2010 19:52:05 GMT Content-Length: 0 <------------> [Jul 6 15:52:05] DEBUG[2782] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.0.5.196:5060 [Jul 6 15:52:05] DEBUG[2689] devicestate.c: No provider found, checking channel drivers for SIP - 33i [Jul 6 15:52:05] DEBUG[2689] chan_sip.c: Checking device state for peer 33i [Jul 6 15:52:05] DEBUG[2689] devicestate.c: Changing state for SIP/33i - state 1 (Not in use) [Jul 6 15:52:05] DEBUG[2689] devicestate.c: device 'SIP/33i' state '1' [Jul 6 15:52:05] DEBUG[2779] app_queue.c: Device 'SIP/33i' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Jul 6 15:52:05] VERBOSE[2782] chan_sip.c: Scheduling destruction of SIP dialog 'c8ce19c6966b23d9' in 32000 ms (Method: REGISTER) [Jul 6 15:52:05] VERBOSE[2782] chan_sip.c: <--- SIP read from UDP:10.0.5.198:5060 ---> REGISTER sip:10.0.5.191 SIP/2.0 Via: SIP/2.0/UDP 10.0.5.198;branch=z9hG4bKc7b5b19dC06BBECC From: "0004f215aabb-1" ;tag=6F5D75EC-D185443D To: CSeq: 9 REGISTER Call-ID: ad2a1218-82f9b90a-393cb63@10.0.5.198 Contact: ;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER" User-Agent: PolycomSoundPointIP-SPIP_501-UA/3.1.6.0017 Accept-Language: fr-fr,fr;q=0.9,en;q=0.8 Authorization: Digest username="0004f215aabb-1", realm="asterisk", nonce="2b1cb8b2", uri="sip:10.0.5.191", response="a0493b32293e6857d24cde8461a0a200", algorithm=MD5 Max-Forwards: 70 Expires: 60 Content-Length: 0 <-------------> [Jul 6 15:52:05] DEBUG[2782] chan_sip.c: Header 0 [ 31]: REGISTER sip:10.0.5.191 SIP/2.0 [Jul 6 15:52:05] DEBUG[2782] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 10.0.5.198;branch=z9hG4bKc7b5b19dC06BBECC [Jul 6 15:52:05] DEBUG[2782] chan_sip.c: Header 2 [ 76]: From: "0004f215aabb-1" ;tag=6F5D75EC-D185443D [Jul 6 15:52:05] DEBUG[2782] chan_sip.c: Header 3 [ 35]: To: [Jul 6 15:52:05] DEBUG[2782] chan_sip.c: Header 4 [ 16]: CSeq: 9 REGISTER [Jul 6 15:52:05] DEBUG[2782] chan_sip.c: Header 5 [ 45]: Call-ID: ad2a1218-82f9b90a-393cb63@10.0.5.198 [Jul 6 15:52:05] DEBUG[2782] chan_sip.c: Header 6 [140]: Contact: ;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER" [Jul 6 15:52:05] DEBUG[2782] chan_sip.c: Header 7 [ 54]: User-Agent: PolycomSoundPointIP-SPIP_501-UA/3.1.6.0017 [Jul 6 15:52:05] DEBUG[2782] chan_sip.c: Header 8 [ 40]: Accept-Language: fr-fr,fr;q=0.9,en;q=0.8 [Jul 6 15:52:05] DEBUG[2782] chan_sip.c: Header 9 [165]: Authorization: Digest username="0004f215aabb-1", realm="asterisk", nonce="2b1cb8b2", uri="sip:10.0.5.191", response="a0493b32293e6857d24cde8461a0a200", algorithm=MD5 [Jul 6 15:52:05] DEBUG[2782] chan_sip.c: Header 10 [ 16]: Max-Forwards: 70 [Jul 6 15:52:05] DEBUG[2782] chan_sip.c: Header 11 [ 11]: Expires: 60 [Jul 6 15:52:05] DEBUG[2782] chan_sip.c: Header 12 [ 17]: Content-Length: 0 [Jul 6 15:52:05] DEBUG[2782] chan_sip.c: Header 13 [ 0]: [Jul 6 15:52:05] VERBOSE[2782] chan_sip.c: --- (13 headers 0 lines) --- [Jul 6 15:52:05] DEBUG[2782] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Jul 6 15:52:05] DEBUG[2782] chan_sip.c: Initializing initreq for method REGISTER - callid ad2a1218-82f9b90a-393cb63@10.0.5.198 [Jul 6 15:52:05] VERBOSE[2782] chan_sip.c: Sending to 10.0.5.198 : 5060 (no NAT) [Jul 6 15:52:05] NOTICE[2782] chan_sip.c: Correct auth, but based on stale nonce received from '' [Jul 6 15:52:05] VERBOSE[2782] chan_sip.c: <--- Transmitting (no NAT) to 10.0.5.198:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.0.5.198;branch=z9hG4bKc7b5b19dC06BBECC;received=10.0.5.198 From: "0004f215aabb-1" ;tag=6F5D75EC-D185443D To: ;tag=as543567ce Call-ID: ad2a1218-82f9b90a-393cb63@10.0.5.198 CSeq: 9 REGISTER Server: Asterisk PBX 1.6.2.9 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="33987b61", stale=true Content-Length: 0 <------------> [Jul 6 15:52:05] DEBUG[2782] chan_sip.c: Trying to put 'SIP/2.0 401' onto UDP socket destined for 10.0.5.198:5060 [Jul 6 15:52:05] VERBOSE[2782] chan_sip.c: Scheduling destruction of SIP dialog 'ad2a1218-82f9b90a-393cb63@10.0.5.198' in 32000 ms (Method: REGISTER) [Jul 6 15:52:05] VERBOSE[2782] chan_sip.c: <--- SIP read from UDP:10.0.5.198:5060 ---> REGISTER sip:10.0.5.191 SIP/2.0 Via: SIP/2.0/UDP 10.0.5.198;branch=z9hG4bK4c231c277707771E From: "0004f215aabb-1" ;tag=6F5D75EC-D185443D To: CSeq: 10 REGISTER Call-ID: ad2a1218-82f9b90a-393cb63@10.0.5.198 Contact: ;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER" User-Agent: PolycomSoundPointIP-SPIP_501-UA/3.1.6.0017 Accept-Language: fr-fr,fr;q=0.9,en;q=0.8 Authorization: Digest username="0004f215aabb-1", realm="asterisk", nonce="33987b61", uri="sip:10.0.5.191", response="6e917c45f50f454054781e5ca18413e0", algorithm=MD5 Max-Forwards: 70 Expires: 60 Content-Length: 0 <-------------> [Jul 6 15:52:05] DEBUG[2782] chan_sip.c: Header 0 [ 31]: REGISTER sip:10.0.5.191 SIP/2.0 [Jul 6 15:52:05] DEBUG[2782] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 10.0.5.198;branch=z9hG4bK4c231c277707771E [Jul 6 15:52:05] DEBUG[2782] chan_sip.c: Header 2 [ 76]: From: "0004f215aabb-1" ;tag=6F5D75EC-D185443D [Jul 6 15:52:05] DEBUG[2782] chan_sip.c: Header 3 [ 35]: To: [Jul 6 15:52:05] DEBUG[2782] chan_sip.c: Header 4 [ 17]: CSeq: 10 REGISTER [Jul 6 15:52:05] DEBUG[2782] chan_sip.c: Header 5 [ 45]: Call-ID: ad2a1218-82f9b90a-393cb63@10.0.5.198 [Jul 6 15:52:05] DEBUG[2782] chan_sip.c: Header 6 [140]: Contact: ;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER" [Jul 6 15:52:05] DEBUG[2782] chan_sip.c: Header 7 [ 54]: User-Agent: PolycomSoundPointIP-SPIP_501-UA/3.1.6.0017 [Jul 6 15:52:05] DEBUG[2782] chan_sip.c: Header 8 [ 40]: Accept-Language: fr-fr,fr;q=0.9,en;q=0.8 [Jul 6 15:52:05] DEBUG[2782] chan_sip.c: Header 9 [165]: Authorization: Digest username="0004f215aabb-1", realm="asterisk", nonce="33987b61", uri="sip:10.0.5.191", response="6e917c45f50f454054781e5ca18413e0", algorithm=MD5 [Jul 6 15:52:05] DEBUG[2782] chan_sip.c: Header 10 [ 16]: Max-Forwards: 70 [Jul 6 15:52:05] DEBUG[2782] chan_sip.c: Header 11 [ 11]: Expires: 60 [Jul 6 15:52:05] DEBUG[2782] chan_sip.c: Header 12 [ 17]: Content-Length: 0 [Jul 6 15:52:05] DEBUG[2782] chan_sip.c: Header 13 [ 0]: [Jul 6 15:52:05] VERBOSE[2782] chan_sip.c: --- (13 headers 0 lines) --- [Jul 6 15:52:05] DEBUG[2782] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Jul 6 15:52:05] DEBUG[2782] chan_sip.c: Initializing initreq for method REGISTER - callid ad2a1218-82f9b90a-393cb63@10.0.5.198 [Jul 6 15:52:05] VERBOSE[2782] chan_sip.c: Sending to 10.0.5.198 : 5060 (no NAT) [Jul 6 15:52:05] VERBOSE[2782] chan_sip.c: <--- Transmitting (no NAT) to 10.0.5.198:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.5.198;branch=z9hG4bK4c231c277707771E;received=10.0.5.198 From: "0004f215aabb-1" ;tag=6F5D75EC-D185443D To: ;tag=as543567ce Call-ID: ad2a1218-82f9b90a-393cb63@10.0.5.198 CSeq: 10 REGISTER Server: Asterisk PBX 1.6.2.9 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Expires: 60 Contact: ;expires=60 Date: Tue, 06 Jul 2010 19:52:05 GMT Content-Length: 0 <------------> [Jul 6 15:52:05] DEBUG[2782] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.0.5.198:5060 [Jul 6 15:52:05] DEBUG[2689] devicestate.c: No provider found, checking channel drivers for SIP - 0004f215aabb-1 [Jul 6 15:52:05] DEBUG[2689] chan_sip.c: Checking device state for peer 0004f215aabb-1 [Jul 6 15:52:05] DEBUG[2689] devicestate.c: Changing state for SIP/0004f215aabb-1 - state 1 (Not in use) [Jul 6 15:52:05] DEBUG[2689] devicestate.c: device 'SIP/0004f215aabb-1' state '1' [Jul 6 15:52:05] DEBUG[2779] app_queue.c: Device 'SIP/0004f215aabb-1' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Jul 6 15:52:05] VERBOSE[2782] chan_sip.c: Scheduling destruction of SIP dialog 'ad2a1218-82f9b90a-393cb63@10.0.5.198' in 32000 ms (Method: REGISTER) [Jul 6 15:52:06] VERBOSE[2782] chan_sip.c: <--- SIP read from UDP:10.0.5.198:5060 ---> REGISTER sip:10.0.5.205 SIP/2.0 Via: SIP/2.0/UDP 10.0.5.198;branch=z9hG4bKcc9551e125584260 From: "0004f215aabb-2" ;tag=9B3D1B49-FDDEE0E6 To: CSeq: 9 REGISTER Call-ID: 5f6fe385-15b70ccf-d62de2d4@10.0.5.198 Contact: ;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER" User-Agent: PolycomSoundPointIP-SPIP_501-UA/3.1.6.0017 Accept-Language: fr-fr,fr;q=0.9,en;q=0.8 Authorization: Digest username="0004f215aabb-2", realm="asterisk", nonce="3b4a9721", uri="sip:10.0.5.205", response="93a1ef6fe61b9ef6126c268cef3a6200", algorithm=MD5 Max-Forwards: 70 Expires: 60 Content-Length: 0 <-------------> [Jul 6 15:52:06] DEBUG[2782] chan_sip.c: Header 0 [ 31]: REGISTER sip:10.0.5.205 SIP/2.0 [Jul 6 15:52:06] DEBUG[2782] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 10.0.5.198;branch=z9hG4bKcc9551e125584260 [Jul 6 15:52:06] DEBUG[2782] chan_sip.c: Header 2 [ 76]: From: "0004f215aabb-2" ;tag=9B3D1B49-FDDEE0E6 [Jul 6 15:52:06] DEBUG[2782] chan_sip.c: Header 3 [ 35]: To: [Jul 6 15:52:06] DEBUG[2782] chan_sip.c: Header 4 [ 16]: CSeq: 9 REGISTER [Jul 6 15:52:06] DEBUG[2782] chan_sip.c: Header 5 [ 46]: Call-ID: 5f6fe385-15b70ccf-d62de2d4@10.0.5.198 [Jul 6 15:52:06] DEBUG[2782] chan_sip.c: Header 6 [140]: Contact: ;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER" [Jul 6 15:52:06] DEBUG[2782] chan_sip.c: Header 7 [ 54]: User-Agent: PolycomSoundPointIP-SPIP_501-UA/3.1.6.0017 [Jul 6 15:52:06] DEBUG[2782] chan_sip.c: Header 8 [ 40]: Accept-Language: fr-fr,fr;q=0.9,en;q=0.8 [Jul 6 15:52:06] DEBUG[2782] chan_sip.c: Header 9 [165]: Authorization: Digest username="0004f215aabb-2", realm="asterisk", nonce="3b4a9721", uri="sip:10.0.5.205", response="93a1ef6fe61b9ef6126c268cef3a6200", algorithm=MD5 [Jul 6 15:52:06] DEBUG[2782] chan_sip.c: Header 10 [ 16]: Max-Forwards: 70 [Jul 6 15:52:06] DEBUG[2782] chan_sip.c: Header 11 [ 11]: Expires: 60 [Jul 6 15:52:06] DEBUG[2782] chan_sip.c: Header 12 [ 17]: Content-Length: 0 [Jul 6 15:52:06] DEBUG[2782] chan_sip.c: Header 13 [ 0]: [Jul 6 15:52:06] VERBOSE[2782] chan_sip.c: --- (13 headers 0 lines) --- [Jul 6 15:52:06] DEBUG[2782] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Jul 6 15:52:06] DEBUG[2782] chan_sip.c: Initializing initreq for method REGISTER - callid 5f6fe385-15b70ccf-d62de2d4@10.0.5.198 [Jul 6 15:52:06] VERBOSE[2782] chan_sip.c: Sending to 10.0.5.198 : 5060 (no NAT) [Jul 6 15:52:06] NOTICE[2782] chan_sip.c: Correct auth, but based on stale nonce received from '' [Jul 6 15:52:06] VERBOSE[2782] chan_sip.c: <--- Transmitting (no NAT) to 10.0.5.198:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.0.5.198;branch=z9hG4bKcc9551e125584260;received=10.0.5.198 From: "0004f215aabb-2" ;tag=9B3D1B49-FDDEE0E6 To: ;tag=as3ebeda14 Call-ID: 5f6fe385-15b70ccf-d62de2d4@10.0.5.198 CSeq: 9 REGISTER Server: Asterisk PBX 1.6.2.9 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="509eb1a0", stale=true Content-Length: 0 <------------> [Jul 6 15:52:06] DEBUG[2782] chan_sip.c: Trying to put 'SIP/2.0 401' onto UDP socket destined for 10.0.5.198:5060 [Jul 6 15:52:06] VERBOSE[2782] chan_sip.c: Scheduling destruction of SIP dialog '5f6fe385-15b70ccf-d62de2d4@10.0.5.198' in 32000 ms (Method: REGISTER) [Jul 6 15:52:06] VERBOSE[2782] chan_sip.c: <--- SIP read from UDP:10.0.5.198:5060 ---> REGISTER sip:10.0.5.205 SIP/2.0 Via: SIP/2.0/UDP 10.0.5.198;branch=z9hG4bK9469d4b15D0FD12 From: "0004f215aabb-2" ;tag=9B3D1B49-FDDEE0E6 To: CSeq: 10 REGISTER Call-ID: 5f6fe385-15b70ccf-d62de2d4@10.0.5.198 Contact: ;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER" User-Agent: PolycomSoundPointIP-SPIP_501-UA/3.1.6.0017 Accept-Language: fr-fr,fr;q=0.9,en;q=0.8 Authorization: Digest username="0004f215aabb-2", realm="asterisk", nonce="509eb1a0", uri="sip:10.0.5.205", response="7327e8f113887adee494a1f66e9f34bc", algorithm=MD5 Max-Forwards: 70 Expires: 60 Content-Length: 0 <-------------> [Jul 6 15:52:06] DEBUG[2782] chan_sip.c: Header 0 [ 31]: REGISTER sip:10.0.5.205 SIP/2.0 [Jul 6 15:52:06] DEBUG[2782] chan_sip.c: Header 1 [ 57]: Via: SIP/2.0/UDP 10.0.5.198;branch=z9hG4bK9469d4b15D0FD12 [Jul 6 15:52:06] DEBUG[2782] chan_sip.c: Header 2 [ 76]: From: "0004f215aabb-2" ;tag=9B3D1B49-FDDEE0E6 [Jul 6 15:52:06] DEBUG[2782] chan_sip.c: Header 3 [ 35]: To: [Jul 6 15:52:06] DEBUG[2782] chan_sip.c: Header 4 [ 17]: CSeq: 10 REGISTER [Jul 6 15:52:06] DEBUG[2782] chan_sip.c: Header 5 [ 46]: Call-ID: 5f6fe385-15b70ccf-d62de2d4@10.0.5.198 [Jul 6 15:52:06] DEBUG[2782] chan_sip.c: Header 6 [140]: Contact: ;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER" [Jul 6 15:52:06] DEBUG[2782] chan_sip.c: Header 7 [ 54]: User-Agent: PolycomSoundPointIP-SPIP_501-UA/3.1.6.0017 [Jul 6 15:52:06] DEBUG[2782] chan_sip.c: Header 8 [ 40]: Accept-Language: fr-fr,fr;q=0.9,en;q=0.8 [Jul 6 15:52:06] DEBUG[2782] chan_sip.c: Header 9 [165]: Authorization: Digest username="0004f215aabb-2", realm="asterisk", nonce="509eb1a0", uri="sip:10.0.5.205", response="7327e8f113887adee494a1f66e9f34bc", algorithm=MD5 [Jul 6 15:52:06] DEBUG[2782] chan_sip.c: Header 10 [ 16]: Max-Forwards: 70 [Jul 6 15:52:06] DEBUG[2782] chan_sip.c: Header 11 [ 11]: Expires: 60 [Jul 6 15:52:06] DEBUG[2782] chan_sip.c: Header 12 [ 17]: Content-Length: 0 [Jul 6 15:52:06] DEBUG[2782] chan_sip.c: Header 13 [ 0]: [Jul 6 15:52:06] VERBOSE[2782] chan_sip.c: --- (13 headers 0 lines) --- [Jul 6 15:52:06] DEBUG[2782] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Jul 6 15:52:06] DEBUG[2782] chan_sip.c: Initializing initreq for method REGISTER - callid 5f6fe385-15b70ccf-d62de2d4@10.0.5.198 [Jul 6 15:52:06] VERBOSE[2782] chan_sip.c: Sending to 10.0.5.198 : 5060 (no NAT) [Jul 6 15:52:06] VERBOSE[2782] chan_sip.c: <--- Transmitting (no NAT) to 10.0.5.198:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.5.198;branch=z9hG4bK9469d4b15D0FD12;received=10.0.5.198 From: "0004f215aabb-2" ;tag=9B3D1B49-FDDEE0E6 To: ;tag=as3ebeda14 Call-ID: 5f6fe385-15b70ccf-d62de2d4@10.0.5.198 CSeq: 10 REGISTER Server: Asterisk PBX 1.6.2.9 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Expires: 60 Contact: ;expires=60 Date: Tue, 06 Jul 2010 19:52:06 GMT Content-Length: 0 <------------> [Jul 6 15:52:06] DEBUG[2782] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.0.5.198:5060 [Jul 6 15:52:06] DEBUG[2689] devicestate.c: No provider found, checking channel drivers for SIP - 0004f215aabb-2 [Jul 6 15:52:06] DEBUG[2689] chan_sip.c: Checking device state for peer 0004f215aabb-2 [Jul 6 15:52:06] DEBUG[2689] devicestate.c: Changing state for SIP/0004f215aabb-2 - state 1 (Not in use) [Jul 6 15:52:06] DEBUG[2689] devicestate.c: device 'SIP/0004f215aabb-2' state '1' [Jul 6 15:52:06] DEBUG[2779] app_queue.c: Device 'SIP/0004f215aabb-2' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Jul 6 15:52:06] VERBOSE[2782] chan_sip.c: Scheduling destruction of SIP dialog '5f6fe385-15b70ccf-d62de2d4@10.0.5.198' in 32000 ms (Method: REGISTER) [Jul 6 15:52:19] VERBOSE[2782] chan_sip.c: <--- SIP read from UDP:10.0.5.196:5060 ---> INVITE sip:901@10.0.5.191:5060 SIP/2.0 Via: SIP/2.0/UDP 10.0.5.196:5060;branch=z9hG4bKbcf2b191673f69e20.605d0de9ef2420970 Max-Forwards: 70 From: "33i" ;tag=8497c4b8bd To: "901" Call-ID: f7bf38449b305573 CSeq: 9031 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Contact: "33i" ;+sip.instance="" Supported: gruu, path, timer, 100rel, replaces User-Agent: Aastra 9143i/2.5.3.2002 Content-Type: application/sdp Content-Length: 612 v=0 o=MxSIP 0 0 IN IP4 10.0.5.196 s=SIP Call c=IN IP4 10.0.5.196 t=0 0 m=audio 3000 RTP/AVP 0 18 106 107 113 110 111 112 98 97 115 96 9 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:106 BV16/8000 a=rtpmap:107 BV32/16000 a=rtpmap:113 L16/16000 a=rtpmap:110 PCMU/16000 a=rtpmap:111 PCMA/16000 a=rtpmap:112 L16/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:115 G726-32/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=silenceSupp:on - - - - a=fmtp:18 annexb=yes a=fmtp:101 0-15 a=ptime:30 a=sendrecv <-------------> [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Header 0 [ 38]: INVITE sip:901@10.0.5.191:5060 SIP/2.0 [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Header 1 [ 82]: Via: SIP/2.0/UDP 10.0.5.196:5060;branch=z9hG4bKbcf2b191673f69e20.605d0de9ef2420970 [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Header 3 [ 52]: From: "33i" ;tag=8497c4b8bd [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Header 4 [ 35]: To: "901" [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Header 5 [ 25]: Call-ID: f7bf38449b305573 [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Header 6 [ 17]: CSeq: 9031 INVITE [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Header 7 [ 88]: Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Header 8 [ 53]: Allow-Events: talk, hold, conference, LocalModeStatus [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Header 9 [118]: Contact: "33i" ;+sip.instance="" [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Header 10 [ 46]: Supported: gruu, path, timer, 100rel, replaces [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Header 11 [ 35]: User-Agent: Aastra 9143i/2.5.3.2002 [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Header 13 [ 19]: Content-Length: 612 [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Header 14 [ 0]: [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Body 0 [ 3]: v=0 [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Body 1 [ 29]: o=MxSIP 0 0 IN IP4 10.0.5.196 [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Body 2 [ 10]: s=SIP Call [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Body 3 [ 19]: c=IN IP4 10.0.5.196 [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Body 4 [ 5]: t=0 0 [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Body 5 [ 70]: m=audio 3000 RTP/AVP 0 18 106 107 113 110 111 112 98 97 115 96 9 8 101 [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000 [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Body 7 [ 21]: a=rtpmap:18 G729/8000 [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Body 8 [ 22]: a=rtpmap:106 BV16/8000 [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Body 9 [ 23]: a=rtpmap:107 BV32/16000 [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Body 10 [ 22]: a=rtpmap:113 L16/16000 [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Body 11 [ 23]: a=rtpmap:110 PCMU/16000 [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Body 12 [ 23]: a=rtpmap:111 PCMA/16000 [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Body 13 [ 21]: a=rtpmap:112 L16/8000 [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Body 14 [ 24]: a=rtpmap:98 G726-16/8000 [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Body 15 [ 24]: a=rtpmap:97 G726-24/8000 [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Body 16 [ 25]: a=rtpmap:115 G726-32/8000 [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Body 17 [ 24]: a=rtpmap:96 G726-40/8000 [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Body 18 [ 20]: a=rtpmap:9 G722/8000 [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Body 19 [ 20]: a=rtpmap:8 PCMA/8000 [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Body 20 [ 33]: a=rtpmap:101 telephone-event/8000 [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Body 21 [ 24]: a=silenceSupp:on - - - - [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Body 22 [ 20]: a=fmtp:18 annexb=yes [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Body 23 [ 15]: a=fmtp:101 0-15 [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Body 24 [ 10]: a=ptime:30 [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Body 25 [ 10]: a=sendrecv [Jul 6 15:52:19] VERBOSE[2782] chan_sip.c: --- (14 headers 26 lines) --- [Jul 6 15:52:19] DEBUG[2782] acl.c: Found IP address for this socket [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 10.0.5.191:5060 [Jul 6 15:52:19] VERBOSE[2782] netsock.c: == Using SIP RTP CoS mark 5 [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Setting NAT on RTP to Off [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Allocating new SIP dialog for f7bf38449b305573 - INVITE (With RTP) [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Begin: parsing SIP "Supported: gruu, path, timer, 100rel, replaces" [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Found SIP option: -gruu- [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Matched SIP option: gruu [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Found SIP option: -path- [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Matched SIP option: path [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Found SIP option: -timer- [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Matched SIP option: timer [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Found SIP option: -100rel- [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Matched SIP option: 100rel [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Found SIP option: -replaces- [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Matched SIP option: replaces [Jul 6 15:52:19] VERBOSE[2782] chan_sip.c: Sending to 10.0.5.196 : 5060 (no NAT) [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Initializing initreq for method INVITE - callid f7bf38449b305573 [Jul 6 15:52:19] VERBOSE[2782] chan_sip.c: Using INVITE request as basis request - f7bf38449b305573 [Jul 6 15:52:19] VERBOSE[2782] chan_sip.c: Found peer '33i' for '33i' from 10.0.5.196:5060 [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Setting NAT on RTP to Off [Jul 6 15:52:19] VERBOSE[2782] chan_sip.c: <--- Reliably Transmitting (no NAT) to 10.0.5.196:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.0.5.196:5060;branch=z9hG4bKbcf2b191673f69e20.605d0de9ef2420970;received=10.0.5.196 From: "33i" ;tag=8497c4b8bd To: "901" ;tag=as45fb5382 Call-ID: f7bf38449b305573 CSeq: 9031 INVITE Server: Asterisk PBX 1.6.2.9 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="05793f28" Content-Length: 0 <------------> [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #54 [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Trying to put 'SIP/2.0 401' onto UDP socket destined for 10.0.5.196:5060 [Jul 6 15:52:19] VERBOSE[2782] chan_sip.c: Scheduling destruction of SIP dialog 'f7bf38449b305573' in 32000 ms (Method: INVITE) [Jul 6 15:52:19] VERBOSE[2782] chan_sip.c: <--- SIP read from UDP:10.0.5.196:5060 ---> ACK sip:901@10.0.5.191:5060 SIP/2.0 Via: SIP/2.0/UDP 10.0.5.196:5060;branch=z9hG4bKbcf2b191673f69e20.605d0de9ef2420970 Max-Forwards: 70 From: "33i" ;tag=8497c4b8bd To: "901" ;tag=as45fb5382 Call-ID: f7bf38449b305573 CSeq: 9031 ACK User-Agent: Aastra 9143i/2.5.3.2002 Content-Length: 0 <-------------> [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Header 0 [ 35]: ACK sip:901@10.0.5.191:5060 SIP/2.0 [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Header 1 [ 82]: Via: SIP/2.0/UDP 10.0.5.196:5060;branch=z9hG4bKbcf2b191673f69e20.605d0de9ef2420970 [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Header 3 [ 52]: From: "33i" ;tag=8497c4b8bd [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Header 4 [ 50]: To: "901" ;tag=as45fb5382 [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Header 5 [ 25]: Call-ID: f7bf38449b305573 [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Header 6 [ 14]: CSeq: 9031 ACK [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Header 7 [ 35]: User-Agent: Aastra 9143i/2.5.3.2002 [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Header 8 [ 17]: Content-Length: 0 [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Header 9 [ 0]: [Jul 6 15:52:19] VERBOSE[2782] chan_sip.c: --- (9 headers 0 lines) --- [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #54 [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Stopping retransmission on 'f7bf38449b305573' of Response 9031: Match Found [Jul 6 15:52:19] VERBOSE[2782] chan_sip.c: <--- SIP read from UDP:10.0.5.196:5060 ---> INVITE sip:901@10.0.5.191:5060 SIP/2.0 Via: SIP/2.0/UDP 10.0.5.196:5060;branch=z9hG4bK2377e9424a4798539.1026cc91a278a4172 Max-Forwards: 70 From: "33i" ;tag=8497c4b8bd To: "901" Call-ID: f7bf38449b305573 CSeq: 9032 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Authorization: Digest username="33i",realm="asterisk",nonce="05793f28",uri="sip:901@10.0.5.191:5060",response="f28bfd3dbfe7d0c1f950665da84a922b",algorithm=MD5 Contact: "33i" ;+sip.instance="" Supported: gruu, path, timer, 100rel, replaces User-Agent: Aastra 9143i/2.5.3.2002 Content-Type: application/sdp Content-Length: 612 v=0 o=MxSIP 0 0 IN IP4 10.0.5.196 s=SIP Call c=IN IP4 10.0.5.196 t=0 0 m=audio 3000 RTP/AVP 0 18 106 107 113 110 111 112 98 97 115 96 9 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:106 BV16/8000 a=rtpmap:107 BV32/16000 a=rtpmap:113 L16/16000 a=rtpmap:110 PCMU/16000 a=rtpmap:111 PCMA/16000 a=rtpmap:112 L16/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:115 G726-32/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=silenceSupp:on - - - - a=fmtp:18 annexb=yes a=fmtp:101 0-15 a=ptime:30 a=sendrecv <-------------> [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Header 0 [ 38]: INVITE sip:901@10.0.5.191:5060 SIP/2.0 [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Header 1 [ 82]: Via: SIP/2.0/UDP 10.0.5.196:5060;branch=z9hG4bK2377e9424a4798539.1026cc91a278a4172 [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Header 3 [ 52]: From: "33i" ;tag=8497c4b8bd [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Header 4 [ 35]: To: "901" [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Header 5 [ 25]: Call-ID: f7bf38449b305573 [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Header 6 [ 17]: CSeq: 9032 INVITE [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Header 7 [ 88]: Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Header 8 [ 53]: Allow-Events: talk, hold, conference, LocalModeStatus [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Header 9 [158]: Authorization: Digest username="33i",realm="asterisk",nonce="05793f28",uri="sip:901@10.0.5.191:5060",response="f28bfd3dbfe7d0c1f950665da84a922b",algorithm=MD5 [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Header 10 [118]: Contact: "33i" ;+sip.instance="" [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Header 11 [ 46]: Supported: gruu, path, timer, 100rel, replaces [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Header 12 [ 35]: User-Agent: Aastra 9143i/2.5.3.2002 [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Header 13 [ 29]: Content-Type: application/sdp [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Header 14 [ 19]: Content-Length: 612 [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Header 15 [ 0]: [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Body 0 [ 3]: v=0 [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Body 1 [ 29]: o=MxSIP 0 0 IN IP4 10.0.5.196 [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Body 2 [ 10]: s=SIP Call [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Body 3 [ 19]: c=IN IP4 10.0.5.196 [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Body 4 [ 5]: t=0 0 [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Body 5 [ 70]: m=audio 3000 RTP/AVP 0 18 106 107 113 110 111 112 98 97 115 96 9 8 101 [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000 [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Body 7 [ 21]: a=rtpmap:18 G729/8000 [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Body 8 [ 22]: a=rtpmap:106 BV16/8000 [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Body 9 [ 23]: a=rtpmap:107 BV32/16000 [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Body 10 [ 22]: a=rtpmap:113 L16/16000 [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Body 11 [ 23]: a=rtpmap:110 PCMU/16000 [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Body 12 [ 23]: a=rtpmap:111 PCMA/16000 [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Body 13 [ 21]: a=rtpmap:112 L16/8000 [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Body 14 [ 24]: a=rtpmap:98 G726-16/8000 [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Body 15 [ 24]: a=rtpmap:97 G726-24/8000 [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Body 16 [ 25]: a=rtpmap:115 G726-32/8000 [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Body 17 [ 24]: a=rtpmap:96 G726-40/8000 [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Body 18 [ 20]: a=rtpmap:9 G722/8000 [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Body 19 [ 20]: a=rtpmap:8 PCMA/8000 [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Body 20 [ 33]: a=rtpmap:101 telephone-event/8000 [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Body 21 [ 24]: a=silenceSupp:on - - - - [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Body 22 [ 20]: a=fmtp:18 annexb=yes [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Body 23 [ 15]: a=fmtp:101 0-15 [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Body 24 [ 10]: a=ptime:30 [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Body 25 [ 10]: a=sendrecv [Jul 6 15:52:19] VERBOSE[2782] chan_sip.c: --- (15 headers 26 lines) --- [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [Jul 6 15:52:19] VERBOSE[2782] chan_sip.c: Sending to 10.0.5.196 : 5060 (no NAT) [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Initializing initreq for method INVITE - callid f7bf38449b305573 [Jul 6 15:52:19] VERBOSE[2782] chan_sip.c: Using INVITE request as basis request - f7bf38449b305573 [Jul 6 15:52:19] VERBOSE[2782] chan_sip.c: Found peer '33i' for '33i' from 10.0.5.196:5060 [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Setting NAT on RTP to Off [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Processing session-level SDP o=MxSIP 0 0 IN IP4 10.0.5.196... UNSUPPORTED. [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Processing session-level SDP s=SIP Call... UNSUPPORTED. [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Processing session-level SDP c=IN IP4 10.0.5.196... OK. [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [Jul 6 15:52:19] VERBOSE[2782] chan_sip.c: Found RTP audio format 0 [Jul 6 15:52:19] VERBOSE[2782] chan_sip.c: Found RTP audio format 18 [Jul 6 15:52:19] VERBOSE[2782] chan_sip.c: Found RTP audio format 106 [Jul 6 15:52:19] VERBOSE[2782] chan_sip.c: Found RTP audio format 107 [Jul 6 15:52:19] VERBOSE[2782] chan_sip.c: Found RTP audio format 113 [Jul 6 15:52:19] VERBOSE[2782] chan_sip.c: Found RTP audio format 110 [Jul 6 15:52:19] VERBOSE[2782] chan_sip.c: Found RTP audio format 111 [Jul 6 15:52:19] VERBOSE[2782] chan_sip.c: Found RTP audio format 112 [Jul 6 15:52:19] VERBOSE[2782] chan_sip.c: Found RTP audio format 98 [Jul 6 15:52:19] VERBOSE[2782] chan_sip.c: Found RTP audio format 97 [Jul 6 15:52:19] VERBOSE[2782] chan_sip.c: Found RTP audio format 115 [Jul 6 15:52:19] VERBOSE[2782] chan_sip.c: Found RTP audio format 96 [Jul 6 15:52:19] VERBOSE[2782] chan_sip.c: Found RTP audio format 9 [Jul 6 15:52:19] VERBOSE[2782] chan_sip.c: Found RTP audio format 8 [Jul 6 15:52:19] VERBOSE[2782] chan_sip.c: Found RTP audio format 101 [Jul 6 15:52:19] VERBOSE[2782] chan_sip.c: Found audio description format PCMU for ID 0 [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Jul 6 15:52:19] VERBOSE[2782] chan_sip.c: Found audio description format G729 for ID 18 [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:18 G729/8000... OK. [Jul 6 15:52:19] VERBOSE[2782] chan_sip.c: Found audio description format BV16 for ID 106 [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:106 BV16/8000... OK. [Jul 6 15:52:19] VERBOSE[2782] chan_sip.c: Found audio description format BV32 for ID 107 [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:107 BV32/16000... OK. [Jul 6 15:52:19] VERBOSE[2782] chan_sip.c: Found audio description format L16 for ID 113 [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:113 L16/16000... OK. [Jul 6 15:52:19] VERBOSE[2782] chan_sip.c: Found audio description format PCMU for ID 110 [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:110 PCMU/16000... OK. [Jul 6 15:52:19] VERBOSE[2782] chan_sip.c: Found audio description format PCMA for ID 111 [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:111 PCMA/16000... OK. [Jul 6 15:52:19] VERBOSE[2782] chan_sip.c: Found audio description format L16 for ID 112 [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:112 L16/8000... OK. [Jul 6 15:52:19] VERBOSE[2782] chan_sip.c: Found audio description format G726-16 for ID 98 [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:98 G726-16/8000... OK. [Jul 6 15:52:19] VERBOSE[2782] chan_sip.c: Found audio description format G726-24 for ID 97 [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:97 G726-24/8000... OK. [Jul 6 15:52:19] VERBOSE[2782] chan_sip.c: Found audio description format G726-32 for ID 115 [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:115 G726-32/8000... OK. [Jul 6 15:52:19] VERBOSE[2782] chan_sip.c: Found audio description format G726-40 for ID 96 [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:96 G726-40/8000... OK. [Jul 6 15:52:19] VERBOSE[2782] chan_sip.c: Found audio description format G722 for ID 9 [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:9 G722/8000... OK. [Jul 6 15:52:19] VERBOSE[2782] chan_sip.c: Found audio description format PCMA for ID 8 [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Jul 6 15:52:19] VERBOSE[2782] chan_sip.c: Found audio description format telephone-event for ID 101 [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Processing media-level (audio) SDP a=silenceSupp:on - - - -... UNSUPPORTED. [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Processing media-level (audio) SDP a=fmtp:18 annexb=yes... UNSUPPORTED. [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED. [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Processing media-level (audio) SDP a=ptime:30... OK. [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Jul 6 15:52:19] VERBOSE[2782] chan_sip.c: Capabilities: us - 0x4 (ulaw), peer - audio=0x8101f4c (ulaw|alaw|g726|slin|g729|speex|ilbc|g722|h263p|t140)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) [Jul 6 15:52:19] VERBOSE[2782] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Jul 6 15:52:19] VERBOSE[2782] chan_sip.c: Peer audio RTP is at port 10.0.5.196:3000 [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: We're settling with these formats: 0x4 (ulaw) [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Checking SIP call limits for device 33i [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Updating call counter for incoming call [Jul 6 15:52:19] VERBOSE[2782] chan_sip.c: Looking for 901 in from-sip (domain 10.0.5.191) [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: *** Our native formats are 0x4 (ulaw) [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: *** Joint capabilities are 0x4 (ulaw) [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: *** Our capabilities are 0x4 (ulaw) [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x4 (ulaw) [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: This channel will not be able to handle video. [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: build_route: Contact hop: "33i" ;+sip.instance="" [Jul 6 15:52:19] VERBOSE[2782] chan_sip.c: list_route: hop: [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Incoming INVITE with 'timer' option enabled [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Session timer started: 56 - f7bf38449b305573 [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: SIP/33i-00000000: New call is still down.... Trying... [Jul 6 15:52:19] VERBOSE[2782] chan_sip.c: <--- Transmitting (no NAT) to 10.0.5.196:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.0.5.196:5060;branch=z9hG4bK2377e9424a4798539.1026cc91a278a4172;received=10.0.5.196 From: "33i" ;tag=8497c4b8bd To: "901" Call-ID: f7bf38449b305573 CSeq: 9032 INVITE Server: Asterisk PBX 1.6.2.9 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Require: timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <------------> [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Trying to put 'SIP/2.0 100' onto UDP socket destined for 10.0.5.196:5060 [Jul 6 15:52:19] DEBUG[2689] devicestate.c: No provider found, checking channel drivers for SIP - 33i [Jul 6 15:52:19] DEBUG[2689] chan_sip.c: Checking device state for peer 33i [Jul 6 15:52:19] DEBUG[2689] devicestate.c: Changing state for SIP/33i - state 1 (Not in use) [Jul 6 15:52:19] DEBUG[2689] devicestate.c: device 'SIP/33i' state '1' [Jul 6 15:52:19] DEBUG[2907] pbx.c: Launching 'Dial' [Jul 6 15:52:19] VERBOSE[2907] pbx.c: -- Executing [901@from-sip:1] Dial("SIP/33i-00000000", "SIP/0004f215aabb-1") in new stack [Jul 6 15:52:19] DEBUG[2907] chan_sip.c: Asked to create a SIP channel with formats: 0x4 (ulaw) [Jul 6 15:52:19] VERBOSE[2907] netsock.c: == Using SIP RTP CoS mark 5 [Jul 6 15:52:19] DEBUG[2907] chan_sip.c: Allocating new SIP dialog for 19d7c4306c89fb7f3ff184e87969b0f0@127.0.0.1 - INVITE (With RTP) [Jul 6 15:52:19] DEBUG[2907] chan_sip.c: Setting NAT on RTP to Off [Jul 6 15:52:19] DEBUG[2907] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Jul 6 15:52:19] DEBUG[2907] acl.c: Found IP address for this socket [Jul 6 15:52:19] DEBUG[2907] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 10.0.5.191:5060 [Jul 6 15:52:19] DEBUG[2907] chan_sip.c: *** Our native formats are 0x4 (ulaw) [Jul 6 15:52:19] DEBUG[2907] chan_sip.c: *** Joint capabilities are 0x4 (ulaw) [Jul 6 15:52:19] DEBUG[2907] chan_sip.c: *** Our capabilities are 0x4 (ulaw) [Jul 6 15:52:19] DEBUG[2907] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x4 (ulaw) [Jul 6 15:52:19] DEBUG[2907] chan_sip.c: *** Our preferred formats from the incoming channel are 0x4 (ulaw) [Jul 6 15:52:19] DEBUG[2907] chan_sip.c: This channel will not be able to handle video. [Jul 6 15:52:19] DEBUG[2907] rtp.c: Seeded SDP of 'SIP/0004f215aabb-1-00000001' with that of 'SIP/33i-00000000' [Jul 6 15:52:19] DEBUG[2907] channel.c: Not copying variable DIALEDTIME. [Jul 6 15:52:19] DEBUG[2907] channel.c: Not copying variable ANSWEREDTIME. [Jul 6 15:52:19] DEBUG[2907] channel.c: Not copying variable DIALEDPEERNAME. [Jul 6 15:52:19] DEBUG[2907] channel.c: Not copying variable DIALEDPEERNUMBER. [Jul 6 15:52:19] DEBUG[2907] channel.c: Not copying variable DIALSTATUS. [Jul 6 15:52:19] DEBUG[2907] channel.c: Not copying variable SIPCALLID. [Jul 6 15:52:19] DEBUG[2907] channel.c: Not copying variable SIPDOMAIN. [Jul 6 15:52:19] DEBUG[2907] channel.c: Not copying variable SIPURI. [Jul 6 15:52:19] DEBUG[2907] chan_sip.c: Outgoing Call for test [Jul 6 15:52:19] DEBUG[2907] chan_sip.c: Updating call counter for outgoing call [Jul 6 15:52:19] DEBUG[2907] chan_sip.c: ** Our capability: 0x4 (ulaw) Video flag: False Text flag: False [Jul 6 15:52:19] DEBUG[2907] chan_sip.c: ** Our prefcodec: 0x4 (ulaw) [Jul 6 15:52:19] VERBOSE[2907] chan_sip.c: Audio is at 10.0.5.191 port 17044 [Jul 6 15:52:19] VERBOSE[2907] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [Jul 6 15:52:19] VERBOSE[2907] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Jul 6 15:52:19] DEBUG[2907] chan_sip.c: -- Done with adding codecs to SDP [Jul 6 15:52:19] DEBUG[2907] chan_sip.c: Done building SDP. Settling with this capability: 0x4 (ulaw) [Jul 6 15:52:19] DEBUG[2907] chan_sip.c: Initializing initreq for method INVITE - callid 4bdb216169a15be718a35d1f6247c7b8@10.0.5.191 [Jul 6 15:52:19] DEBUG[2907] chan_sip.c: Header 0 [ 44]: INVITE sip:0004f215aabb-1@10.0.5.198 SIP/2.0 [Jul 6 15:52:19] DEBUG[2907] chan_sip.c: Header 1 [ 61]: Via: SIP/2.0/UDP 10.0.5.191:5060;branch=z9hG4bK3bc2c42f;rport [Jul 6 15:52:19] DEBUG[2907] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jul 6 15:52:19] DEBUG[2907] chan_sip.c: Header 3 [ 47]: From: "33i" ;tag=as5a66f1a4 [Jul 6 15:52:19] DEBUG[2907] chan_sip.c: Header 4 [ 35]: To: [Jul 6 15:52:19] DEBUG[2907] chan_sip.c: Header 5 [ 29]: Contact: [Jul 6 15:52:19] DEBUG[2907] chan_sip.c: Header 6 [ 52]: Call-ID: 4bdb216169a15be718a35d1f6247c7b8@10.0.5.191 [Jul 6 15:52:19] DEBUG[2907] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Jul 6 15:52:19] DEBUG[2907] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 1.6.2.9 [Jul 6 15:52:19] DEBUG[2907] chan_sip.c: Header 9 [ 35]: Date: Tue, 06 Jul 2010 19:52:19 GMT [Jul 6 15:52:19] DEBUG[2907] chan_sip.c: Header 10 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Jul 6 15:52:19] DEBUG[2907] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Jul 6 15:52:19] DEBUG[2907] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Jul 6 15:52:19] DEBUG[2907] chan_sip.c: Header 13 [ 19]: Content-Length: 230 [Jul 6 15:52:19] DEBUG[2907] chan_sip.c: Header 14 [ 0]: [Jul 6 15:52:19] DEBUG[2907] chan_sip.c: Body 0 [ 3]: v=0 [Jul 6 15:52:19] DEBUG[2907] chan_sip.c: Body 1 [ 44]: o=root 740359899 740359899 IN IP4 10.0.5.191 [Jul 6 15:52:19] DEBUG[2907] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 1.6.2.9 [Jul 6 15:52:19] DEBUG[2907] chan_sip.c: Body 3 [ 19]: c=IN IP4 10.0.5.191 [Jul 6 15:52:19] DEBUG[2907] chan_sip.c: Body 4 [ 5]: t=0 0 [Jul 6 15:52:19] DEBUG[2907] chan_sip.c: Body 5 [ 27]: m=audio 17044 RTP/AVP 0 101 [Jul 6 15:52:19] DEBUG[2907] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000 [Jul 6 15:52:19] DEBUG[2907] chan_sip.c: Body 7 [ 33]: a=rtpmap:101 telephone-event/8000 [Jul 6 15:52:19] DEBUG[2907] chan_sip.c: Body 8 [ 15]: a=fmtp:101 0-16 [Jul 6 15:52:19] DEBUG[2907] chan_sip.c: Body 9 [ 10]: a=ptime:20 [Jul 6 15:52:19] DEBUG[2907] chan_sip.c: Body 10 [ 10]: a=sendrecv [Jul 6 15:52:19] VERBOSE[2907] chan_sip.c: Reliably Transmitting (no NAT) to 10.0.5.198:5060: INVITE sip:0004f215aabb-1@10.0.5.198 SIP/2.0 Via: SIP/2.0/UDP 10.0.5.191:5060;branch=z9hG4bK3bc2c42f;rport Max-Forwards: 70 From: "33i" ;tag=as5a66f1a4 To: Contact: Call-ID: 4bdb216169a15be718a35d1f6247c7b8@10.0.5.191 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.2.9 Date: Tue, 06 Jul 2010 19:52:19 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 230 v=0 o=root 740359899 740359899 IN IP4 10.0.5.191 s=Asterisk PBX 1.6.2.9 c=IN IP4 10.0.5.191 t=0 0 m=audio 17044 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- [Jul 6 15:52:19] DEBUG[2907] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #58 [Jul 6 15:52:19] DEBUG[2907] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 10.0.5.198:5060 [Jul 6 15:52:19] VERBOSE[2907] app_dial.c: -- Called 0004f215aabb-1 [Jul 6 15:52:19] DEBUG[2779] app_queue.c: Device 'SIP/33i' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Jul 6 15:52:19] VERBOSE[2782] chan_sip.c: <--- SIP read from UDP:10.0.5.198:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.0.5.191:5060;branch=z9hG4bK3bc2c42f;rport From: "33i" ;tag=as5a66f1a4 To: ;tag=5B8DFB2F-A3F8FFB4 CSeq: 102 INVITE Call-ID: 4bdb216169a15be718a35d1f6247c7b8@10.0.5.191 Contact: User-Agent: PolycomSoundPointIP-SPIP_501-UA/3.1.6.0017 Accept-Language: fr-fr,fr;q=0.9,en;q=0.8 Content-Length: 0 <-------------> [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Header 1 [ 61]: Via: SIP/2.0/UDP 10.0.5.191:5060;branch=z9hG4bK3bc2c42f;rport [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Header 2 [ 47]: From: "33i" ;tag=as5a66f1a4 [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Header 3 [ 57]: To: ;tag=5B8DFB2F-A3F8FFB4 [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Header 4 [ 16]: CSeq: 102 INVITE [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Header 5 [ 52]: Call-ID: 4bdb216169a15be718a35d1f6247c7b8@10.0.5.191 [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Header 6 [ 40]: Contact: [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Header 7 [ 54]: User-Agent: PolycomSoundPointIP-SPIP_501-UA/3.1.6.0017 [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Header 8 [ 40]: Accept-Language: fr-fr,fr;q=0.9,en;q=0.8 [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Header 10 [ 0]: [Jul 6 15:52:19] VERBOSE[2782] chan_sip.c: --- (10 headers 0 lines) --- [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: *** SIP TIMER: Cancelling retransmission #58 - INVITE (got response) [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '4bdb216169a15be718a35d1f6247c7b8@10.0.5.191' Request 102: Found [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: SIP response 100 to standard invite [Jul 6 15:52:19] VERBOSE[2782] chan_sip.c: <--- SIP read from UDP:10.0.5.198:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.0.5.191:5060;branch=z9hG4bK3bc2c42f;rport From: "33i" ;tag=as5a66f1a4 To: ;tag=5B8DFB2F-A3F8FFB4 CSeq: 102 INVITE Call-ID: 4bdb216169a15be718a35d1f6247c7b8@10.0.5.191 Contact: User-Agent: PolycomSoundPointIP-SPIP_501-UA/3.1.6.0017 Allow-Events: talk,hold,conference Accept-Language: fr-fr,fr;q=0.9,en;q=0.8 Content-Length: 0 <-------------> [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Header 0 [ 19]: SIP/2.0 180 Ringing [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Header 1 [ 61]: Via: SIP/2.0/UDP 10.0.5.191:5060;branch=z9hG4bK3bc2c42f;rport [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Header 2 [ 47]: From: "33i" ;tag=as5a66f1a4 [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Header 3 [ 57]: To: ;tag=5B8DFB2F-A3F8FFB4 [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Header 4 [ 16]: CSeq: 102 INVITE [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Header 5 [ 52]: Call-ID: 4bdb216169a15be718a35d1f6247c7b8@10.0.5.191 [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Header 6 [ 40]: Contact: [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Header 7 [ 54]: User-Agent: PolycomSoundPointIP-SPIP_501-UA/3.1.6.0017 [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Header 8 [ 34]: Allow-Events: talk,hold,conference [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Header 9 [ 40]: Accept-Language: fr-fr,fr;q=0.9,en;q=0.8 [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: Header 11 [ 0]: [Jul 6 15:52:19] VERBOSE[2782] chan_sip.c: --- (11 headers 0 lines) --- [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '4bdb216169a15be718a35d1f6247c7b8@10.0.5.191' Request 102: Found [Jul 6 15:52:19] DEBUG[2782] chan_sip.c: SIP response 180 to standard invite [Jul 6 15:52:19] VERBOSE[2907] app_dial.c: -- SIP/0004f215aabb-1-00000001 is ringing [Jul 6 15:52:19] DEBUG[2907] rtp.c: Setting early bridge SDP of 'SIP/33i-00000000' with that of 'SIP/0004f215aabb-1-00000001' [Jul 6 15:52:19] VERBOSE[2907] chan_sip.c: <--- Transmitting (no NAT) to 10.0.5.196:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.0.5.196:5060;branch=z9hG4bK2377e9424a4798539.1026cc91a278a4172;received=10.0.5.196 From: "33i" ;tag=8497c4b8bd To: "901" ;tag=as6ef584fe Call-ID: f7bf38449b305573 CSeq: 9032 INVITE Server: Asterisk PBX 1.6.2.9 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Require: timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <------------> [Jul 6 15:52:19] DEBUG[2907] chan_sip.c: Trying to put 'SIP/2.0 180' onto UDP socket destined for 10.0.5.196:5060 [Jul 6 15:52:19] DEBUG[2689] devicestate.c: No provider found, checking channel drivers for SIP - 0004f215aabb-1 [Jul 6 15:52:19] DEBUG[2689] chan_sip.c: Checking device state for peer 0004f215aabb-1 [Jul 6 15:52:19] DEBUG[2689] devicestate.c: Changing state for SIP/0004f215aabb-1 - state 1 (Not in use) [Jul 6 15:52:19] DEBUG[2689] devicestate.c: device 'SIP/0004f215aabb-1' state '1' [Jul 6 15:52:19] DEBUG[2779] app_queue.c: Device 'SIP/0004f215aabb-1' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Jul 6 15:52:21] VERBOSE[2782] chan_sip.c: <--- SIP read from UDP:10.0.5.198:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.5.191:5060;branch=z9hG4bK3bc2c42f;rport From: "33i" ;tag=as5a66f1a4 To: ;tag=5B8DFB2F-A3F8FFB4 CSeq: 102 INVITE Call-ID: 4bdb216169a15be718a35d1f6247c7b8@10.0.5.191 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_501-UA/3.1.6.0017 Accept-Language: fr-fr,fr;q=0.9,en;q=0.8 Content-Type: application/sdp Content-Length: 195 v=0 o=- 1278441041 1278441041 IN IP4 10.0.5.198 s=Polycom IP Phone c=IN IP4 10.0.5.198 t=0 0 m=audio 2258 RTP/AVP 0 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 <-------------> [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Header 1 [ 61]: Via: SIP/2.0/UDP 10.0.5.191:5060;branch=z9hG4bK3bc2c42f;rport [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Header 2 [ 47]: From: "33i" ;tag=as5a66f1a4 [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Header 3 [ 57]: To: ;tag=5B8DFB2F-A3F8FFB4 [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Header 4 [ 16]: CSeq: 102 INVITE [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Header 5 [ 52]: Call-ID: 4bdb216169a15be718a35d1f6247c7b8@10.0.5.191 [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Header 6 [ 40]: Contact: [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Header 7 [ 96]: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Header 8 [ 54]: User-Agent: PolycomSoundPointIP-SPIP_501-UA/3.1.6.0017 [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Header 9 [ 40]: Accept-Language: fr-fr,fr;q=0.9,en;q=0.8 [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Header 10 [ 29]: Content-Type: application/sdp [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Header 11 [ 19]: Content-Length: 195 [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Header 12 [ 0]: [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Body 0 [ 3]: v=0 [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Body 1 [ 43]: o=- 1278441041 1278441041 IN IP4 10.0.5.198 [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Body 2 [ 18]: s=Polycom IP Phone [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Body 3 [ 19]: c=IN IP4 10.0.5.198 [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Body 4 [ 5]: t=0 0 [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Body 5 [ 26]: m=audio 2258 RTP/AVP 0 101 [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Body 6 [ 10]: a=sendrecv [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 PCMU/8000 [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Jul 6 15:52:21] VERBOSE[2782] chan_sip.c: --- (12 headers 9 lines) --- [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Acked pending invite 102 [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Stopping retransmission on '4bdb216169a15be718a35d1f6247c7b8@10.0.5.191' of Request 102: Match Found [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: SIP response 200 to standard invite [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Processing session-level SDP o=- 1278441041 1278441041 IN IP4 10.0.5.198... UNSUPPORTED. [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Processing session-level SDP s=Polycom IP Phone... UNSUPPORTED. [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Processing session-level SDP c=IN IP4 10.0.5.198... OK. [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [Jul 6 15:52:21] VERBOSE[2782] chan_sip.c: Found RTP audio format 0 [Jul 6 15:52:21] VERBOSE[2782] chan_sip.c: Found RTP audio format 101 [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Jul 6 15:52:21] VERBOSE[2782] chan_sip.c: Found audio description format PCMU for ID 0 [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Jul 6 15:52:21] VERBOSE[2782] chan_sip.c: Found audio description format telephone-event for ID 101 [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Jul 6 15:52:21] VERBOSE[2782] chan_sip.c: Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) [Jul 6 15:52:21] VERBOSE[2782] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Jul 6 15:52:21] VERBOSE[2782] chan_sip.c: Peer audio RTP is at port 10.0.5.198:2258 [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: We're settling with these formats: 0x4 (ulaw) [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: We have an owner, now see if we need to change this call [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Updating call counter for outgoing call [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: build_route: Contact hop: [Jul 6 15:52:21] VERBOSE[2782] chan_sip.c: list_route: hop: [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Strict routing enforced for session 4bdb216169a15be718a35d1f6247c7b8@10.0.5.191 [Jul 6 15:52:21] VERBOSE[2782] chan_sip.c: set_destination: Parsing for address/port to send to [Jul 6 15:52:21] VERBOSE[2782] chan_sip.c: set_destination: set destination to 10.0.5.198, port 5060 [Jul 6 15:52:21] VERBOSE[2782] chan_sip.c: Transmitting (no NAT) to 10.0.5.198:5060: ACK sip:0004f215aabb-1@10.0.5.198 SIP/2.0 Via: SIP/2.0/UDP 10.0.5.191:5060;branch=z9hG4bK184f4cfa;rport Max-Forwards: 70 From: "33i" ;tag=as5a66f1a4 To: ;tag=5B8DFB2F-A3F8FFB4 Contact: Call-ID: 4bdb216169a15be718a35d1f6247c7b8@10.0.5.191 CSeq: 102 ACK User-Agent: Asterisk PBX 1.6.2.9 Content-Length: 0 --- [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Trying to put 'ACK sip:000' onto UDP socket destined for 10.0.5.198:5060 [Jul 6 15:52:21] VERBOSE[2907] app_dial.c: -- SIP/0004f215aabb-1-00000001 answered SIP/33i-00000000 [Jul 6 15:52:21] DEBUG[2907] rtp.c: Setting early bridge SDP of 'SIP/33i-00000000' with that of 'SIP/0004f215aabb-1-00000001' [Jul 6 15:52:21] DEBUG[2907] chan_sip.c: SIP answering channel: SIP/33i-00000000 [Jul 6 15:52:21] DEBUG[2907] rtp.c: Setting the marker bit due to a source update [Jul 6 15:52:21] DEBUG[2907] chan_sip.c: Setting framing from config on incoming call [Jul 6 15:52:21] DEBUG[2907] chan_sip.c: ** Our capability: 0x4 (ulaw) Video flag: True Text flag: True [Jul 6 15:52:21] DEBUG[2907] chan_sip.c: ** Our prefcodec: 0x0 (nothing) [Jul 6 15:52:21] VERBOSE[2907] chan_sip.c: Audio is at 10.0.5.191 port 19436 [Jul 6 15:52:21] VERBOSE[2907] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [Jul 6 15:52:21] VERBOSE[2907] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Jul 6 15:52:21] DEBUG[2907] chan_sip.c: -- Done with adding codecs to SDP [Jul 6 15:52:21] DEBUG[2907] chan_sip.c: Done building SDP. Settling with this capability: 0x4 (ulaw) [Jul 6 15:52:21] VERBOSE[2907] chan_sip.c: <--- Reliably Transmitting (no NAT) to 10.0.5.196:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.5.196:5060;branch=z9hG4bK2377e9424a4798539.1026cc91a278a4172;received=10.0.5.196 From: "33i" ;tag=8497c4b8bd To: "901" ;tag=as6ef584fe Call-ID: f7bf38449b305573 CSeq: 9032 INVITE Server: Asterisk PBX 1.6.2.9 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Require: timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Content-Length: 230 v=0 o=root 578486317 578486317 IN IP4 10.0.5.191 s=Asterisk PBX 1.6.2.9 c=IN IP4 10.0.5.191 t=0 0 m=audio 19436 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <------------> [Jul 6 15:52:21] DEBUG[2907] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #61 [Jul 6 15:52:21] DEBUG[2907] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.0.5.196:5060 [Jul 6 15:52:21] DEBUG[2907] features.c: bridge answer set, chan answer set [Jul 6 15:52:21] DEBUG[2907] rtp.c: Setting the marker bit due to a source update [Jul 6 15:52:21] DEBUG[2907] rtp.c: Setting the marker bit due to a source update [Jul 6 15:52:21] VERBOSE[2907] rtp.c: -- Native bridging SIP/33i-00000000 and SIP/0004f215aabb-1-00000001 [Jul 6 15:52:21] DEBUG[2907] chan_sip.c: Deferring reinvite on SIP 'f7bf38449b305573' - It's audio will be redirected to IP 10.0.5.198 [Jul 6 15:52:21] DEBUG[2907] chan_sip.c: Sending reinvite on SIP '4bdb216169a15be718a35d1f6247c7b8@10.0.5.191' - It's audio soon redirected to IP 10.0.5.196 [Jul 6 15:52:21] DEBUG[2907] chan_sip.c: Strict routing enforced for session 4bdb216169a15be718a35d1f6247c7b8@10.0.5.191 [Jul 6 15:52:21] VERBOSE[2907] chan_sip.c: set_destination: Parsing for address/port to send to [Jul 6 15:52:21] VERBOSE[2907] chan_sip.c: set_destination: set destination to 10.0.5.198, port 5060 [Jul 6 15:52:21] DEBUG[2907] chan_sip.c: ** Our capability: 0x4 (ulaw) Video flag: True Text flag: True [Jul 6 15:52:21] DEBUG[2907] chan_sip.c: ** Our prefcodec: 0x4 (ulaw) [Jul 6 15:52:21] VERBOSE[2907] chan_sip.c: Audio is at 10.0.5.191 port 17044 [Jul 6 15:52:21] VERBOSE[2907] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [Jul 6 15:52:21] VERBOSE[2907] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Jul 6 15:52:21] DEBUG[2907] chan_sip.c: -- Done with adding codecs to SDP [Jul 6 15:52:21] DEBUG[2907] chan_sip.c: Done building SDP. Settling with this capability: 0x4 (ulaw) [Jul 6 15:52:21] DEBUG[2907] chan_sip.c: Initializing already initialized SIP dialog 4bdb216169a15be718a35d1f6247c7b8@10.0.5.191 (presumably reinvite) [Jul 6 15:52:21] DEBUG[2907] chan_sip.c: Header 0 [ 44]: INVITE sip:0004f215aabb-1@10.0.5.198 SIP/2.0 [Jul 6 15:52:21] DEBUG[2907] chan_sip.c: Header 1 [ 61]: Via: SIP/2.0/UDP 10.0.5.191:5060;branch=z9hG4bK4344009e;rport [Jul 6 15:52:21] DEBUG[2907] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jul 6 15:52:21] DEBUG[2907] chan_sip.c: Header 3 [ 47]: From: "33i" ;tag=as5a66f1a4 [Jul 6 15:52:21] DEBUG[2907] chan_sip.c: Header 4 [ 57]: To: ;tag=5B8DFB2F-A3F8FFB4 [Jul 6 15:52:21] DEBUG[2907] chan_sip.c: Header 5 [ 29]: Contact: [Jul 6 15:52:21] DEBUG[2907] chan_sip.c: Header 6 [ 52]: Call-ID: 4bdb216169a15be718a35d1f6247c7b8@10.0.5.191 [Jul 6 15:52:21] DEBUG[2907] chan_sip.c: Header 7 [ 16]: CSeq: 103 INVITE [Jul 6 15:52:21] DEBUG[2907] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 1.6.2.9 [Jul 6 15:52:21] DEBUG[2907] chan_sip.c: Header 9 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Jul 6 15:52:21] DEBUG[2907] chan_sip.c: Header 10 [ 26]: Supported: replaces, timer [Jul 6 15:52:21] DEBUG[2907] chan_sip.c: Header 11 [ 52]: X-asterisk-Info: SIP re-invite (External RTP bridge) [Jul 6 15:52:21] DEBUG[2907] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Jul 6 15:52:21] DEBUG[2907] chan_sip.c: Header 13 [ 19]: Content-Length: 229 [Jul 6 15:52:21] DEBUG[2907] chan_sip.c: Header 14 [ 0]: [Jul 6 15:52:21] DEBUG[2907] chan_sip.c: Body 0 [ 3]: v=0 [Jul 6 15:52:21] DEBUG[2907] chan_sip.c: Body 1 [ 44]: o=root 740359899 740359900 IN IP4 10.0.5.196 [Jul 6 15:52:21] DEBUG[2907] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 1.6.2.9 [Jul 6 15:52:21] DEBUG[2907] chan_sip.c: Body 3 [ 19]: c=IN IP4 10.0.5.196 [Jul 6 15:52:21] DEBUG[2907] chan_sip.c: Body 4 [ 5]: t=0 0 [Jul 6 15:52:21] DEBUG[2907] chan_sip.c: Body 5 [ 26]: m=audio 3000 RTP/AVP 0 101 [Jul 6 15:52:21] DEBUG[2907] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000 [Jul 6 15:52:21] DEBUG[2907] chan_sip.c: Body 7 [ 33]: a=rtpmap:101 telephone-event/8000 [Jul 6 15:52:21] DEBUG[2907] chan_sip.c: Body 8 [ 15]: a=fmtp:101 0-16 [Jul 6 15:52:21] DEBUG[2907] chan_sip.c: Body 9 [ 10]: a=ptime:20 [Jul 6 15:52:21] DEBUG[2907] chan_sip.c: Body 10 [ 10]: a=sendrecv [Jul 6 15:52:21] VERBOSE[2907] chan_sip.c: Reliably Transmitting (no NAT) to 10.0.5.198:5060: INVITE sip:0004f215aabb-1@10.0.5.198 SIP/2.0 Via: SIP/2.0/UDP 10.0.5.191:5060;branch=z9hG4bK4344009e;rport Max-Forwards: 70 From: "33i" ;tag=as5a66f1a4 To: ;tag=5B8DFB2F-A3F8FFB4 Contact: Call-ID: 4bdb216169a15be718a35d1f6247c7b8@10.0.5.191 CSeq: 103 INVITE User-Agent: Asterisk PBX 1.6.2.9 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 229 v=0 o=root 740359899 740359900 IN IP4 10.0.5.196 s=Asterisk PBX 1.6.2.9 c=IN IP4 10.0.5.196 t=0 0 m=audio 3000 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- [Jul 6 15:52:21] DEBUG[2907] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #62 [Jul 6 15:52:21] DEBUG[2907] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 10.0.5.198:5060 [Jul 6 15:52:21] DEBUG[2689] devicestate.c: No provider found, checking channel drivers for SIP - 0004f215aabb-1 [Jul 6 15:52:21] DEBUG[2689] chan_sip.c: Checking device state for peer 0004f215aabb-1 [Jul 6 15:52:21] DEBUG[2689] devicestate.c: Changing state for SIP/0004f215aabb-1 - state 1 (Not in use) [Jul 6 15:52:21] DEBUG[2689] devicestate.c: device 'SIP/0004f215aabb-1' state '1' [Jul 6 15:52:21] DEBUG[2689] devicestate.c: No provider found, checking channel drivers for SIP - 33i [Jul 6 15:52:21] DEBUG[2689] chan_sip.c: Checking device state for peer 33i [Jul 6 15:52:21] DEBUG[2689] devicestate.c: Changing state for SIP/33i - state 1 (Not in use) [Jul 6 15:52:21] DEBUG[2689] devicestate.c: device 'SIP/33i' state '1' [Jul 6 15:52:21] DEBUG[2779] app_queue.c: Device 'SIP/0004f215aabb-1' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Jul 6 15:52:21] DEBUG[2779] app_queue.c: Device 'SIP/33i' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Jul 6 15:52:21] VERBOSE[2782] chan_sip.c: <--- SIP read from UDP:10.0.5.196:5060 ---> ACK sip:901@10.0.5.191 SIP/2.0 Via: SIP/2.0/UDP 10.0.5.196:5060;branch=z9hG4bK652b8a04ba9b6390e.0cf0e56649d069e44 Max-Forwards: 70 From: "33i" ;tag=8497c4b8bd To: "901" ;tag=as6ef584fe Call-ID: f7bf38449b305573 CSeq: 9032 ACK Authorization: Digest username="33i",realm="asterisk",nonce="05793f28",uri="sip:901@10.0.5.191:5060",response="f28bfd3dbfe7d0c1f950665da84a922b",algorithm=MD5 User-Agent: Aastra 9143i/2.5.3.2002 Content-Length: 0 <-------------> [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Header 0 [ 30]: ACK sip:901@10.0.5.191 SIP/2.0 [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Header 1 [ 82]: Via: SIP/2.0/UDP 10.0.5.196:5060;branch=z9hG4bK652b8a04ba9b6390e.0cf0e56649d069e44 [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Header 3 [ 52]: From: "33i" ;tag=8497c4b8bd [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Header 4 [ 50]: To: "901" ;tag=as6ef584fe [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Header 5 [ 25]: Call-ID: f7bf38449b305573 [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Header 6 [ 14]: CSeq: 9032 ACK [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Header 7 [158]: Authorization: Digest username="33i",realm="asterisk",nonce="05793f28",uri="sip:901@10.0.5.191:5060",response="f28bfd3dbfe7d0c1f950665da84a922b",algorithm=MD5 [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Header 8 [ 35]: User-Agent: Aastra 9143i/2.5.3.2002 [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Header 10 [ 0]: [Jul 6 15:52:21] VERBOSE[2782] chan_sip.c: --- (10 headers 0 lines) --- [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #61 [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Stopping retransmission on 'f7bf38449b305573' of Response 9032: Match Found [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Sending pending reinvite on 'f7bf38449b305573' [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Strict routing enforced for session f7bf38449b305573 [Jul 6 15:52:21] VERBOSE[2782] chan_sip.c: set_destination: Parsing for address/port to send to [Jul 6 15:52:21] VERBOSE[2782] chan_sip.c: set_destination: set destination to 10.0.5.196, port 5060 [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: ** Our capability: 0x4 (ulaw) Video flag: True Text flag: True [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: ** Our prefcodec: 0x0 (nothing) [Jul 6 15:52:21] VERBOSE[2782] chan_sip.c: Audio is at 10.0.5.191 port 19436 [Jul 6 15:52:21] VERBOSE[2782] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [Jul 6 15:52:21] VERBOSE[2782] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: -- Done with adding codecs to SDP [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Done building SDP. Settling with this capability: 0x4 (ulaw) [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Initializing already initialized SIP dialog f7bf38449b305573 (presumably reinvite) [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Header 0 [ 52]: INVITE sip:33i@10.0.5.196:5060;transport=udp SIP/2.0 [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Header 1 [ 61]: Via: SIP/2.0/UDP 10.0.5.191:5060;branch=z9hG4bK1eabe87b;rport [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Header 3 [ 52]: From: "901" ;tag=as6ef584fe [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Header 4 [ 50]: To: "33i" ;tag=8497c4b8bd [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Header 5 [ 29]: Contact: [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Header 6 [ 25]: Call-ID: f7bf38449b305573 [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 1.6.2.9 [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Header 9 [ 14]: Require: timer [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Header 10 [ 35]: Session-Expires: 1800;refresher=uas [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Header 11 [ 10]: Min-SE: 90 [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Header 12 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Header 13 [ 26]: Supported: replaces, timer [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Header 14 [ 52]: X-asterisk-Info: SIP re-invite (External RTP bridge) [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Header 15 [ 29]: Content-Type: application/sdp [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Header 16 [ 19]: Content-Length: 229 [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Header 17 [ 0]: [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Body 0 [ 3]: v=0 [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Body 1 [ 44]: o=root 578486317 578486318 IN IP4 10.0.5.198 [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 1.6.2.9 [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Body 3 [ 19]: c=IN IP4 10.0.5.198 [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Body 4 [ 5]: t=0 0 [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Body 5 [ 26]: m=audio 2258 RTP/AVP 0 101 [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000 [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Body 7 [ 33]: a=rtpmap:101 telephone-event/8000 [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Body 8 [ 15]: a=fmtp:101 0-16 [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Body 9 [ 10]: a=ptime:20 [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Body 10 [ 10]: a=sendrecv [Jul 6 15:52:21] VERBOSE[2782] chan_sip.c: Reliably Transmitting (no NAT) to 10.0.5.196:5060: INVITE sip:33i@10.0.5.196:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.0.5.191:5060;branch=z9hG4bK1eabe87b;rport Max-Forwards: 70 From: "901" ;tag=as6ef584fe To: "33i" ;tag=8497c4b8bd Contact: Call-ID: f7bf38449b305573 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.2.9 Require: timer Session-Expires: 1800;refresher=uas Min-SE: 90 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 229 v=0 o=root 578486317 578486318 IN IP4 10.0.5.198 s=Asterisk PBX 1.6.2.9 c=IN IP4 10.0.5.198 t=0 0 m=audio 2258 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #63 [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 10.0.5.196:5060 [Jul 6 15:52:21] DEBUG[2907] rtp.c: Ooh, format changed from unknown to ulaw [Jul 6 15:52:21] DEBUG[2907] rtp.c: Created smoother: format: 4 ms: 20 len: 160 [Jul 6 15:52:21] VERBOSE[2782] chan_sip.c: <--- SIP read from UDP:10.0.5.198:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.5.191:5060;branch=z9hG4bK4344009e;rport From: "33i" ;tag=as5a66f1a4 To: ;tag=5B8DFB2F-A3F8FFB4 CSeq: 103 INVITE Call-ID: 4bdb216169a15be718a35d1f6247c7b8@10.0.5.191 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_501-UA/3.1.6.0017 Accept-Language: fr-fr,fr;q=0.9,en;q=0.8 Content-Type: application/sdp Content-Length: 195 v=0 o=- 1278441041 1278441042 IN IP4 10.0.5.198 s=Polycom IP Phone c=IN IP4 10.0.5.198 t=0 0 m=audio 2258 RTP/AVP 0 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 <-------------> [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Header 1 [ 61]: Via: SIP/2.0/UDP 10.0.5.191:5060;branch=z9hG4bK4344009e;rport [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Header 2 [ 47]: From: "33i" ;tag=as5a66f1a4 [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Header 3 [ 57]: To: ;tag=5B8DFB2F-A3F8FFB4 [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Header 4 [ 16]: CSeq: 103 INVITE [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Header 5 [ 52]: Call-ID: 4bdb216169a15be718a35d1f6247c7b8@10.0.5.191 [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Header 6 [ 40]: Contact: [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Header 7 [ 96]: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Header 8 [ 54]: User-Agent: PolycomSoundPointIP-SPIP_501-UA/3.1.6.0017 [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Header 9 [ 40]: Accept-Language: fr-fr,fr;q=0.9,en;q=0.8 [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Header 10 [ 29]: Content-Type: application/sdp [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Header 11 [ 19]: Content-Length: 195 [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Header 12 [ 0]: [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Body 0 [ 3]: v=0 [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Body 1 [ 43]: o=- 1278441041 1278441042 IN IP4 10.0.5.198 [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Body 2 [ 18]: s=Polycom IP Phone [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Body 3 [ 19]: c=IN IP4 10.0.5.198 [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Body 4 [ 5]: t=0 0 [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Body 5 [ 26]: m=audio 2258 RTP/AVP 0 101 [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Body 6 [ 10]: a=sendrecv [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 PCMU/8000 [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Jul 6 15:52:21] VERBOSE[2782] chan_sip.c: --- (12 headers 9 lines) --- [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Acked pending invite 103 [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #62 [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Stopping retransmission on '4bdb216169a15be718a35d1f6247c7b8@10.0.5.191' of Request 103: Match Found [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: SIP response 200 to RE-invite on outgoing call 4bdb216169a15be718a35d1f6247c7b8@10.0.5.191 [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Processing session-level SDP o=- 1278441041 1278441042 IN IP4 10.0.5.198... UNSUPPORTED. [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Processing session-level SDP s=Polycom IP Phone... UNSUPPORTED. [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Processing session-level SDP c=IN IP4 10.0.5.198... OK. [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [Jul 6 15:52:21] VERBOSE[2782] chan_sip.c: Found RTP audio format 0 [Jul 6 15:52:21] VERBOSE[2782] chan_sip.c: Found RTP audio format 101 [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Jul 6 15:52:21] VERBOSE[2782] chan_sip.c: Found audio description format PCMU for ID 0 [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Jul 6 15:52:21] VERBOSE[2782] chan_sip.c: Found audio description format telephone-event for ID 101 [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Jul 6 15:52:21] VERBOSE[2782] chan_sip.c: Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) [Jul 6 15:52:21] VERBOSE[2782] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Jul 6 15:52:21] VERBOSE[2782] chan_sip.c: Peer audio RTP is at port 10.0.5.198:2258 [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: We're settling with these formats: 0x4 (ulaw) [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: We have an owner, now see if we need to change this call [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Updating call counter for outgoing call [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Strict routing enforced for session 4bdb216169a15be718a35d1f6247c7b8@10.0.5.191 [Jul 6 15:52:21] VERBOSE[2782] chan_sip.c: set_destination: Parsing for address/port to send to [Jul 6 15:52:21] VERBOSE[2782] chan_sip.c: set_destination: set destination to 10.0.5.198, port 5060 [Jul 6 15:52:21] VERBOSE[2782] chan_sip.c: Transmitting (no NAT) to 10.0.5.198:5060: ACK sip:0004f215aabb-1@10.0.5.198 SIP/2.0 Via: SIP/2.0/UDP 10.0.5.191:5060;branch=z9hG4bK6ee0fbf7;rport Max-Forwards: 70 From: "33i" ;tag=as5a66f1a4 To: ;tag=5B8DFB2F-A3F8FFB4 Contact: Call-ID: 4bdb216169a15be718a35d1f6247c7b8@10.0.5.191 CSeq: 103 ACK User-Agent: Asterisk PBX 1.6.2.9 Content-Length: 0 --- [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Trying to put 'ACK sip:000' onto UDP socket destined for 10.0.5.198:5060 [Jul 6 15:52:21] DEBUG[2907] rtp.c: Ooh, format changed from unknown to ulaw [Jul 6 15:52:21] DEBUG[2907] rtp.c: Created smoother: format: 4 ms: 20 len: 160 [Jul 6 15:52:21] VERBOSE[2782] chan_sip.c: <--- SIP read from UDP:10.0.5.196:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.5.191:5060;branch=z9hG4bK1eabe87b;rport=5060;received=10.0.5.191 From: "901" ;tag=as6ef584fe To: "33i" ;tag=8497c4b8bd Call-ID: f7bf38449b305573 CSeq: 102 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Contact: "33i" ;+sip.instance="" Require: timer Server: Aastra 9143i/2.5.3.2002 Session-Expires: 1800;refresher=uas Supported: gruu, path, timer, replaces Content-Type: application/sdp Content-Length: 202 v=0 o=MxSIP 0 1 IN IP4 10.0.5.196 s=SIP Call c=IN IP4 10.0.5.196 t=0 0 m=audio 3000 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv <-------------> [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Header 1 [ 86]: Via: SIP/2.0/UDP 10.0.5.191:5060;branch=z9hG4bK1eabe87b;rport=5060;received=10.0.5.191 [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Header 2 [ 52]: From: "901" ;tag=as6ef584fe [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Header 3 [ 50]: To: "33i" ;tag=8497c4b8bd [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Header 4 [ 25]: Call-ID: f7bf38449b305573 [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Header 6 [ 88]: Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Header 7 [ 53]: Allow-Events: talk, hold, conference, LocalModeStatus [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Header 8 [118]: Contact: "33i" ;+sip.instance="" [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Header 9 [ 14]: Require: timer [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Header 10 [ 31]: Server: Aastra 9143i/2.5.3.2002 [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Header 11 [ 35]: Session-Expires: 1800;refresher=uas [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Header 12 [ 38]: Supported: gruu, path, timer, replaces [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Header 13 [ 29]: Content-Type: application/sdp [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Header 14 [ 19]: Content-Length: 202 [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Header 15 [ 0]: [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Body 0 [ 3]: v=0 [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Body 1 [ 29]: o=MxSIP 0 1 IN IP4 10.0.5.196 [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Body 2 [ 10]: s=SIP Call [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Body 3 [ 19]: c=IN IP4 10.0.5.196 [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Body 4 [ 5]: t=0 0 [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Body 5 [ 26]: m=audio 3000 RTP/AVP 0 101 [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000 [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Body 7 [ 33]: a=rtpmap:101 telephone-event/8000 [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Body 8 [ 15]: a=fmtp:101 0-15 [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Body 9 [ 10]: a=ptime:20 [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Body 10 [ 10]: a=sendrecv [Jul 6 15:52:21] VERBOSE[2782] chan_sip.c: --- (15 headers 11 lines) --- [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Acked pending invite 102 [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #63 [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Stopping retransmission on 'f7bf38449b305573' of Request 102: Match Found [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: SIP response 200 to RE-invite on outgoing call f7bf38449b305573 [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Processing session-level SDP o=MxSIP 0 1 IN IP4 10.0.5.196... UNSUPPORTED. [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Processing session-level SDP s=SIP Call... UNSUPPORTED. [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Processing session-level SDP c=IN IP4 10.0.5.196... OK. [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [Jul 6 15:52:21] VERBOSE[2782] chan_sip.c: Found RTP audio format 0 [Jul 6 15:52:21] VERBOSE[2782] chan_sip.c: Found RTP audio format 101 [Jul 6 15:52:21] VERBOSE[2782] chan_sip.c: Found audio description format PCMU for ID 0 [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Jul 6 15:52:21] VERBOSE[2782] chan_sip.c: Found audio description format telephone-event for ID 101 [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED. [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK. [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Jul 6 15:52:21] VERBOSE[2782] chan_sip.c: Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) [Jul 6 15:52:21] VERBOSE[2782] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Jul 6 15:52:21] VERBOSE[2782] chan_sip.c: Peer audio RTP is at port 10.0.5.196:3000 [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: We're settling with these formats: 0x4 (ulaw) [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: We have an owner, now see if we need to change this call [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Updating call counter for incoming call [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Strict routing enforced for session f7bf38449b305573 [Jul 6 15:52:21] VERBOSE[2782] chan_sip.c: set_destination: Parsing for address/port to send to [Jul 6 15:52:21] VERBOSE[2782] chan_sip.c: set_destination: set destination to 10.0.5.196, port 5060 [Jul 6 15:52:21] VERBOSE[2782] chan_sip.c: Transmitting (no NAT) to 10.0.5.196:5060: ACK sip:33i@10.0.5.196:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.0.5.191:5060;branch=z9hG4bK08b81469;rport Max-Forwards: 70 From: "901" ;tag=as6ef584fe To: "33i" ;tag=8497c4b8bd Contact: Call-ID: f7bf38449b305573 CSeq: 102 ACK User-Agent: Asterisk PBX 1.6.2.9 Content-Length: 0 --- [Jul 6 15:52:21] DEBUG[2782] chan_sip.c: Trying to put 'ACK sip:33i' onto UDP socket destined for 10.0.5.196:5060 [Jul 6 15:52:22] VERBOSE[2782] chan_sip.c: <--- SIP read from UDP:10.0.5.198:5060 ---> INVITE sip:33i@10.0.5.191 SIP/2.0 Via: SIP/2.0/UDP 10.0.5.198;branch=z9hG4bK7b753ec6B63FDEA9 From: ;tag=5B8DFB2F-A3F8FFB4 To: "33i" ;tag=as5a66f1a4 CSeq: 1 INVITE Call-ID: 4bdb216169a15be718a35d1f6247c7b8@10.0.5.191 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_501-UA/3.1.6.0017 Accept-Language: fr-fr,fr;q=0.9,en;q=0.8 Supported: 100rel,replaces Allow-Events: talk,hold,conference Max-Forwards: 70 Content-Type: application/sdp Content-Length: 207 v=0 o=- 1278441041 1278441043 IN IP4 10.0.5.198 s=Polycom IP Phone c=IN IP4 10.0.5.198 t=0 0 a=sendonly m=audio 2258 RTP/AVP 0 101 a=sendonly a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 <-------------> [Jul 6 15:52:22] DEBUG[2782] chan_sip.c: Header 0 [ 33]: INVITE sip:33i@10.0.5.191 SIP/2.0 [Jul 6 15:52:22] DEBUG[2782] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 10.0.5.198;branch=z9hG4bK7b753ec6B63FDEA9 [Jul 6 15:52:22] DEBUG[2782] chan_sip.c: Header 2 [ 59]: From: ;tag=5B8DFB2F-A3F8FFB4 [Jul 6 15:52:22] DEBUG[2782] chan_sip.c: Header 3 [ 45]: To: "33i" ;tag=as5a66f1a4 [Jul 6 15:52:22] DEBUG[2782] chan_sip.c: Header 4 [ 14]: CSeq: 1 INVITE [Jul 6 15:52:22] DEBUG[2782] chan_sip.c: Header 5 [ 52]: Call-ID: 4bdb216169a15be718a35d1f6247c7b8@10.0.5.191 [Jul 6 15:52:22] DEBUG[2782] chan_sip.c: Header 6 [ 40]: Contact: [Jul 6 15:52:22] DEBUG[2782] chan_sip.c: Header 7 [ 96]: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER [Jul 6 15:52:22] DEBUG[2782] chan_sip.c: Header 8 [ 54]: User-Agent: PolycomSoundPointIP-SPIP_501-UA/3.1.6.0017 [Jul 6 15:52:22] DEBUG[2782] chan_sip.c: Header 9 [ 40]: Accept-Language: fr-fr,fr;q=0.9,en;q=0.8 [Jul 6 15:52:22] DEBUG[2782] chan_sip.c: Header 10 [ 26]: Supported: 100rel,replaces [Jul 6 15:52:22] DEBUG[2782] chan_sip.c: Header 11 [ 34]: Allow-Events: talk,hold,conference [Jul 6 15:52:22] DEBUG[2782] chan_sip.c: Header 12 [ 16]: Max-Forwards: 70 [Jul 6 15:52:22] DEBUG[2782] chan_sip.c: Header 13 [ 29]: Content-Type: application/sdp [Jul 6 15:52:22] DEBUG[2782] chan_sip.c: Header 14 [ 19]: Content-Length: 207 [Jul 6 15:52:22] DEBUG[2782] chan_sip.c: Header 15 [ 0]: [Jul 6 15:52:22] DEBUG[2782] chan_sip.c: Body 0 [ 3]: v=0 [Jul 6 15:52:22] DEBUG[2782] chan_sip.c: Body 1 [ 43]: o=- 1278441041 1278441043 IN IP4 10.0.5.198 [Jul 6 15:52:22] DEBUG[2782] chan_sip.c: Body 2 [ 18]: s=Polycom IP Phone [Jul 6 15:52:22] DEBUG[2782] chan_sip.c: Body 3 [ 19]: c=IN IP4 10.0.5.198 [Jul 6 15:52:22] DEBUG[2782] chan_sip.c: Body 4 [ 5]: t=0 0 [Jul 6 15:52:22] DEBUG[2782] chan_sip.c: Body 5 [ 10]: a=sendonly [Jul 6 15:52:22] DEBUG[2782] chan_sip.c: Body 6 [ 26]: m=audio 2258 RTP/AVP 0 101 [Jul 6 15:52:22] DEBUG[2782] chan_sip.c: Body 7 [ 10]: a=sendonly [Jul 6 15:52:22] DEBUG[2782] chan_sip.c: Body 8 [ 20]: a=rtpmap:0 PCMU/8000 [Jul 6 15:52:22] DEBUG[2782] chan_sip.c: Body 9 [ 33]: a=rtpmap:101 telephone-event/8000 [Jul 6 15:52:22] VERBOSE[2782] chan_sip.c: --- (15 headers 10 lines) --- [Jul 6 15:52:22] DEBUG[2782] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [Jul 6 15:52:22] DEBUG[2782] chan_sip.c: Begin: parsing SIP "Supported: 100rel,replaces" [Jul 6 15:52:22] DEBUG[2782] chan_sip.c: Found SIP option: -100rel- [Jul 6 15:52:22] DEBUG[2782] chan_sip.c: Matched SIP option: 100rel [Jul 6 15:52:22] DEBUG[2782] chan_sip.c: Found SIP option: -replaces- [Jul 6 15:52:22] DEBUG[2782] chan_sip.c: Matched SIP option: replaces [Jul 6 15:52:22] VERBOSE[2782] chan_sip.c: Sending to 10.0.5.198 : 5060 (no NAT) [Jul 6 15:52:22] DEBUG[2782] chan_sip.c: Initializing initreq for method INVITE - callid 4bdb216169a15be718a35d1f6247c7b8@10.0.5.191 [Jul 6 15:52:22] DEBUG[2782] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Jul 6 15:52:22] DEBUG[2782] chan_sip.c: Processing session-level SDP o=- 1278441041 1278441043 IN IP4 10.0.5.198... UNSUPPORTED. [Jul 6 15:52:22] DEBUG[2782] chan_sip.c: Processing session-level SDP s=Polycom IP Phone... UNSUPPORTED. [Jul 6 15:52:22] DEBUG[2782] chan_sip.c: Processing session-level SDP c=IN IP4 10.0.5.198... OK. [Jul 6 15:52:22] DEBUG[2782] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [Jul 6 15:52:22] DEBUG[2782] chan_sip.c: Processing session-level SDP a=sendonly... OK. [Jul 6 15:52:22] VERBOSE[2782] chan_sip.c: Found RTP audio format 0 [Jul 6 15:52:22] VERBOSE[2782] chan_sip.c: Found RTP audio format 101 [Jul 6 15:52:22] DEBUG[2782] chan_sip.c: Processing media-level (audio) SDP a=sendonly... OK. [Jul 6 15:52:22] VERBOSE[2782] chan_sip.c: Found audio description format PCMU for ID 0 [Jul 6 15:52:22] DEBUG[2782] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Jul 6 15:52:22] VERBOSE[2782] chan_sip.c: Found audio description format telephone-event for ID 101 [Jul 6 15:52:22] DEBUG[2782] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Jul 6 15:52:22] VERBOSE[2782] chan_sip.c: Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) [Jul 6 15:52:22] VERBOSE[2782] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Jul 6 15:52:22] VERBOSE[2782] chan_sip.c: Peer audio RTP is at port 10.0.5.198:2258 [Jul 6 15:52:22] DEBUG[2782] chan_sip.c: We're settling with these formats: 0x4 (ulaw) [Jul 6 15:52:22] DEBUG[2782] chan_sip.c: We have an owner, now see if we need to change this call [Jul 6 15:52:22] DEBUG[2782] chan_sip.c: Got a SIP re-invite for call 4bdb216169a15be718a35d1f6247c7b8@10.0.5.191 [Jul 6 15:52:22] DEBUG[2782] chan_sip.c: SIP/0004f215aabb-1-00000001: This call is UP.... [Jul 6 15:52:22] VERBOSE[2782] chan_sip.c: <--- Transmitting (no NAT) to 10.0.5.198:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.0.5.198;branch=z9hG4bK7b753ec6B63FDEA9;received=10.0.5.198 From: ;tag=5B8DFB2F-A3F8FFB4 To: "33i" ;tag=as5a66f1a4 Call-ID: 4bdb216169a15be718a35d1f6247c7b8@10.0.5.191 CSeq: 1 INVITE Server: Asterisk PBX 1.6.2.9 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: Content-Length: 0 <------------> [Jul 6 15:52:22] DEBUG[2782] chan_sip.c: Trying to put 'SIP/2.0 100' onto UDP socket destined for 10.0.5.198:5060 [Jul 6 15:52:22] DEBUG[2782] chan_sip.c: Setting framing from config on incoming call [Jul 6 15:52:22] DEBUG[2782] chan_sip.c: ** Our capability: 0x4 (ulaw) Video flag: True Text flag: True [Jul 6 15:52:22] DEBUG[2782] chan_sip.c: ** Our prefcodec: 0x4 (ulaw) [Jul 6 15:52:22] VERBOSE[2782] chan_sip.c: Audio is at 10.0.5.191 port 17044 [Jul 6 15:52:22] VERBOSE[2782] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [Jul 6 15:52:22] VERBOSE[2782] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Jul 6 15:52:22] DEBUG[2782] chan_sip.c: -- Done with adding codecs to SDP [Jul 6 15:52:22] DEBUG[2782] chan_sip.c: Done building SDP. Settling with this capability: 0x4 (ulaw) [Jul 6 15:52:22] VERBOSE[2782] chan_sip.c: <--- Reliably Transmitting (no NAT) to 10.0.5.198:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.5.198;branch=z9hG4bK7b753ec6B63FDEA9;received=10.0.5.198 From: ;tag=5B8DFB2F-A3F8FFB4 To: "33i" ;tag=as5a66f1a4 Call-ID: 4bdb216169a15be718a35d1f6247c7b8@10.0.5.191 CSeq: 1 INVITE Server: Asterisk PBX 1.6.2.9 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 229 v=0 o=root 740359899 740359901 IN IP4 10.0.5.196 s=Asterisk PBX 1.6.2.9 c=IN IP4 10.0.5.196 t=0 0 m=audio 3000 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=recvonly <------------> [Jul 6 15:52:22] DEBUG[2782] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #66 [Jul 6 15:52:22] DEBUG[2782] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.0.5.198:5060 [Jul 6 15:52:22] DEBUG[2907] chan_sip.c: Sending reinvite on SIP 'f7bf38449b305573' - It's audio soon redirected to IP 10.0.5.191 [Jul 6 15:52:22] DEBUG[2907] chan_sip.c: Strict routing enforced for session f7bf38449b305573 [Jul 6 15:52:22] VERBOSE[2907] chan_sip.c: set_destination: Parsing for address/port to send to [Jul 6 15:52:22] VERBOSE[2907] chan_sip.c: set_destination: set destination to 10.0.5.196, port 5060 [Jul 6 15:52:22] DEBUG[2907] chan_sip.c: ** Our capability: 0x4 (ulaw) Video flag: True Text flag: True [Jul 6 15:52:22] DEBUG[2907] chan_sip.c: ** Our prefcodec: 0x0 (nothing) [Jul 6 15:52:22] VERBOSE[2907] chan_sip.c: Audio is at 10.0.5.191 port 19436 [Jul 6 15:52:22] VERBOSE[2907] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [Jul 6 15:52:22] VERBOSE[2907] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Jul 6 15:52:22] DEBUG[2907] chan_sip.c: -- Done with adding codecs to SDP [Jul 6 15:52:22] DEBUG[2907] chan_sip.c: Done building SDP. Settling with this capability: 0x4 (ulaw) [Jul 6 15:52:22] DEBUG[2907] chan_sip.c: Initializing already initialized SIP dialog f7bf38449b305573 (presumably reinvite) [Jul 6 15:52:22] DEBUG[2907] chan_sip.c: Header 0 [ 52]: INVITE sip:33i@10.0.5.196:5060;transport=udp SIP/2.0 [Jul 6 15:52:22] DEBUG[2907] chan_sip.c: Header 1 [ 61]: Via: SIP/2.0/UDP 10.0.5.191:5060;branch=z9hG4bK25aa96cf;rport [Jul 6 15:52:22] DEBUG[2907] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jul 6 15:52:22] DEBUG[2907] chan_sip.c: Header 3 [ 52]: From: "901" ;tag=as6ef584fe [Jul 6 15:52:22] DEBUG[2907] chan_sip.c: Header 4 [ 50]: To: "33i" ;tag=8497c4b8bd [Jul 6 15:52:22] DEBUG[2907] chan_sip.c: Header 5 [ 29]: Contact: [Jul 6 15:52:22] DEBUG[2907] chan_sip.c: Header 6 [ 25]: Call-ID: f7bf38449b305573 [Jul 6 15:52:22] DEBUG[2907] chan_sip.c: Header 7 [ 16]: CSeq: 103 INVITE [Jul 6 15:52:22] DEBUG[2907] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 1.6.2.9 [Jul 6 15:52:22] DEBUG[2907] chan_sip.c: Header 9 [ 14]: Require: timer [Jul 6 15:52:22] DEBUG[2907] chan_sip.c: Header 10 [ 35]: Session-Expires: 1800;refresher=uas [Jul 6 15:52:22] DEBUG[2907] chan_sip.c: Header 11 [ 10]: Min-SE: 90 [Jul 6 15:52:22] DEBUG[2907] chan_sip.c: Header 12 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Jul 6 15:52:22] DEBUG[2907] chan_sip.c: Header 13 [ 26]: Supported: replaces, timer [Jul 6 15:52:22] DEBUG[2907] chan_sip.c: Header 14 [ 52]: X-asterisk-Info: SIP re-invite (External RTP bridge) [Jul 6 15:52:22] DEBUG[2907] chan_sip.c: Header 15 [ 29]: Content-Type: application/sdp [Jul 6 15:52:22] DEBUG[2907] chan_sip.c: Header 16 [ 19]: Content-Length: 230 [Jul 6 15:52:22] DEBUG[2907] chan_sip.c: Header 17 [ 0]: [Jul 6 15:52:22] DEBUG[2907] chan_sip.c: Body 0 [ 3]: v=0 [Jul 6 15:52:22] DEBUG[2907] chan_sip.c: Body 1 [ 44]: o=root 578486317 578486319 IN IP4 10.0.5.191 [Jul 6 15:52:22] DEBUG[2907] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 1.6.2.9 [Jul 6 15:52:22] DEBUG[2907] chan_sip.c: Body 3 [ 19]: c=IN IP4 10.0.5.191 [Jul 6 15:52:22] DEBUG[2907] chan_sip.c: Body 4 [ 5]: t=0 0 [Jul 6 15:52:22] DEBUG[2907] chan_sip.c: Body 5 [ 27]: m=audio 19436 RTP/AVP 0 101 [Jul 6 15:52:22] DEBUG[2907] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000 [Jul 6 15:52:22] DEBUG[2907] chan_sip.c: Body 7 [ 33]: a=rtpmap:101 telephone-event/8000 [Jul 6 15:52:22] DEBUG[2907] chan_sip.c: Body 8 [ 15]: a=fmtp:101 0-16 [Jul 6 15:52:22] DEBUG[2907] chan_sip.c: Body 9 [ 10]: a=ptime:20 [Jul 6 15:52:22] DEBUG[2907] chan_sip.c: Body 10 [ 10]: a=sendrecv [Jul 6 15:52:22] VERBOSE[2907] chan_sip.c: Reliably Transmitting (no NAT) to 10.0.5.196:5060: INVITE sip:33i@10.0.5.196:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.0.5.191:5060;branch=z9hG4bK25aa96cf;rport Max-Forwards: 70 From: "901" ;tag=as6ef584fe To: "33i" ;tag=8497c4b8bd Contact: Call-ID: f7bf38449b305573 CSeq: 103 INVITE User-Agent: Asterisk PBX 1.6.2.9 Require: timer Session-Expires: 1800;refresher=uas Min-SE: 90 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 230 v=0 o=root 578486317 578486319 IN IP4 10.0.5.191 s=Asterisk PBX 1.6.2.9 c=IN IP4 10.0.5.191 t=0 0 m=audio 19436 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- [Jul 6 15:52:22] DEBUG[2907] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #67 [Jul 6 15:52:22] DEBUG[2907] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 10.0.5.196:5060 [Jul 6 15:52:22] DEBUG[2907] rtp.c: Setting the marker bit due to a source update [Jul 6 15:52:22] VERBOSE[2907] res_musiconhold.c: -- Started music on hold, class 'default', on SIP/33i-00000000 [Jul 6 15:52:22] DEBUG[2907] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [Jul 6 15:52:22] DEBUG[2907] rtp.c: Setting the marker bit due to a source update [Jul 6 15:52:22] DEBUG[2907] channel.c: Set channel SIP/33i-00000000 to write format slin [Jul 6 15:52:22] DEBUG[2907] res_musiconhold.c: SIP/33i-00000000 Opened file 1 '/var/lib/asterisk/moh/manolo_camp-morning_coffee' [Jul 6 15:52:22] DEBUG[2907] rtp.c: Difference is 7648, ms is 976 [Jul 6 15:52:22] VERBOSE[2782] chan_sip.c: <--- SIP read from UDP:10.0.5.196:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.5.191:5060;branch=z9hG4bK25aa96cf;rport=5060;received=10.0.5.191 From: "901" ;tag=as6ef584fe To: "33i" ;tag=8497c4b8bd Call-ID: f7bf38449b305573 CSeq: 103 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Contact: "33i" ;+sip.instance="" Require: timer Server: Aastra 9143i/2.5.3.2002 Session-Expires: 1800;refresher=uas Supported: gruu, path, timer, replaces Content-Type: application/sdp Content-Length: 202 v=0 o=MxSIP 0 2 IN IP4 10.0.5.196 s=SIP Call c=IN IP4 10.0.5.196 t=0 0 m=audio 3000 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv <-------------> [Jul 6 15:52:22] DEBUG[2782] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Jul 6 15:52:22] DEBUG[2782] chan_sip.c: Header 1 [ 86]: Via: SIP/2.0/UDP 10.0.5.191:5060;branch=z9hG4bK25aa96cf;rport=5060;received=10.0.5.191 [Jul 6 15:52:22] DEBUG[2782] chan_sip.c: Header 2 [ 52]: From: "901" ;tag=as6ef584fe [Jul 6 15:52:22] DEBUG[2782] chan_sip.c: Header 3 [ 50]: To: "33i" ;tag=8497c4b8bd [Jul 6 15:52:22] DEBUG[2782] chan_sip.c: Header 4 [ 25]: Call-ID: f7bf38449b305573 [Jul 6 15:52:22] DEBUG[2782] chan_sip.c: Header 5 [ 16]: CSeq: 103 INVITE [Jul 6 15:52:22] DEBUG[2782] chan_sip.c: Header 6 [ 88]: Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO [Jul 6 15:52:22] DEBUG[2782] chan_sip.c: Header 7 [ 53]: Allow-Events: talk, hold, conference, LocalModeStatus [Jul 6 15:52:22] DEBUG[2782] chan_sip.c: Header 8 [118]: Contact: "33i" ;+sip.instance="" [Jul 6 15:52:22] DEBUG[2782] chan_sip.c: Header 9 [ 14]: Require: timer [Jul 6 15:52:22] DEBUG[2782] chan_sip.c: Header 10 [ 31]: Server: Aastra 9143i/2.5.3.2002 [Jul 6 15:52:22] DEBUG[2782] chan_sip.c: Header 11 [ 35]: Session-Expires: 1800;refresher=uas [Jul 6 15:52:22] DEBUG[2782] chan_sip.c: Header 12 [ 38]: Supported: gruu, path, timer, replaces [Jul 6 15:52:22] DEBUG[2782] chan_sip.c: Header 13 [ 29]: Content-Type: application/sdp [Jul 6 15:52:22] DEBUG[2782] chan_sip.c: Header 14 [ 19]: Content-Length: 202 [Jul 6 15:52:22] DEBUG[2782] chan_sip.c: Header 15 [ 0]: [Jul 6 15:52:22] DEBUG[2782] chan_sip.c: Body 0 [ 3]: v=0 [Jul 6 15:52:22] DEBUG[2782] chan_sip.c: Body 1 [ 29]: o=MxSIP 0 2 IN IP4 10.0.5.196 [Jul 6 15:52:22] DEBUG[2782] chan_sip.c: Body 2 [ 10]: s=SIP Call [Jul 6 15:52:22] DEBUG[2782] chan_sip.c: Body 3 [ 19]: c=IN IP4 10.0.5.196 [Jul 6 15:52:22] DEBUG[2782] chan_sip.c: Body 4 [ 5]: t=0 0 [Jul 6 15:52:22] DEBUG[2782] chan_sip.c: Body 5 [ 26]: m=audio 3000 RTP/AVP 0 101 [Jul 6 15:52:22] DEBUG[2782] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000 [Jul 6 15:52:22] DEBUG[2782] chan_sip.c: Body 7 [ 33]: a=rtpmap:101 telephone-event/8000 [Jul 6 15:52:22] DEBUG[2782] chan_sip.c: Body 8 [ 15]: a=fmtp:101 0-15 [Jul 6 15:52:22] DEBUG[2782] chan_sip.c: Body 9 [ 10]: a=ptime:20 [Jul 6 15:52:22] DEBUG[2782] chan_sip.c: Body 10 [ 10]: a=sendrecv [Jul 6 15:52:22] VERBOSE[2782] chan_sip.c: --- (15 headers 11 lines) --- [Jul 6 15:52:22] DEBUG[2782] chan_sip.c: Acked pending invite 103 [Jul 6 15:52:22] DEBUG[2782] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #67 [Jul 6 15:52:22] DEBUG[2782] chan_sip.c: Stopping retransmission on 'f7bf38449b305573' of Request 103: Match Found [Jul 6 15:52:22] DEBUG[2782] chan_sip.c: SIP response 200 to RE-invite on outgoing call f7bf38449b305573 [Jul 6 15:52:22] DEBUG[2782] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Jul 6 15:52:22] DEBUG[2782] chan_sip.c: Processing session-level SDP o=MxSIP 0 2 IN IP4 10.0.5.196... UNSUPPORTED. [Jul 6 15:52:22] DEBUG[2782] chan_sip.c: Processing session-level SDP s=SIP Call... UNSUPPORTED. [Jul 6 15:52:22] DEBUG[2782] chan_sip.c: Processing session-level SDP c=IN IP4 10.0.5.196... OK. [Jul 6 15:52:22] DEBUG[2782] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [Jul 6 15:52:22] VERBOSE[2782] chan_sip.c: Found RTP audio format 0 [Jul 6 15:52:22] VERBOSE[2782] chan_sip.c: Found RTP audio format 101 [Jul 6 15:52:22] VERBOSE[2782] chan_sip.c: Found audio description format PCMU for ID 0 [Jul 6 15:52:22] DEBUG[2782] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Jul 6 15:52:22] VERBOSE[2782] chan_sip.c: Found audio description format telephone-event for ID 101 [Jul 6 15:52:22] DEBUG[2782] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Jul 6 15:52:22] DEBUG[2782] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED. [Jul 6 15:52:22] DEBUG[2782] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK. [Jul 6 15:52:22] DEBUG[2782] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Jul 6 15:52:22] VERBOSE[2782] chan_sip.c: Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) [Jul 6 15:52:22] VERBOSE[2782] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Jul 6 15:52:22] VERBOSE[2782] chan_sip.c: Peer audio RTP is at port 10.0.5.196:3000 [Jul 6 15:52:22] DEBUG[2782] chan_sip.c: We're settling with these formats: 0x4 (ulaw) [Jul 6 15:52:22] DEBUG[2782] chan_sip.c: We have an owner, now see if we need to change this call [Jul 6 15:52:22] DEBUG[2782] chan_sip.c: Updating call counter for incoming call [Jul 6 15:52:22] DEBUG[2782] chan_sip.c: Strict routing enforced for session f7bf38449b305573 [Jul 6 15:52:22] VERBOSE[2782] chan_sip.c: set_destination: Parsing for address/port to send to [Jul 6 15:52:22] VERBOSE[2782] chan_sip.c: set_destination: set destination to 10.0.5.196, port 5060 [Jul 6 15:52:22] VERBOSE[2782] chan_sip.c: Transmitting (no NAT) to 10.0.5.196:5060: ACK sip:33i@10.0.5.196:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.0.5.191:5060;branch=z9hG4bK6a9a1bc2;rport Max-Forwards: 70 From: "901" ;tag=as6ef584fe To: "33i" ;tag=8497c4b8bd Contact: Call-ID: f7bf38449b305573 CSeq: 103 ACK User-Agent: Asterisk PBX 1.6.2.9 Content-Length: 0 --- [Jul 6 15:52:22] DEBUG[2782] chan_sip.c: Trying to put 'ACK sip:33i' onto UDP socket destined for 10.0.5.196:5060 [Jul 6 15:52:22] VERBOSE[2782] chan_sip.c: <--- SIP read from UDP:10.0.5.198:5060 ---> ACK sip:33i@10.0.5.191 SIP/2.0 Via: SIP/2.0/UDP 10.0.5.198;branch=z9hG4bK5f206ac8AF82B9D3 From: ;tag=5B8DFB2F-A3F8FFB4 To: "33i" ;tag=as5a66f1a4 CSeq: 1 ACK Call-ID: 4bdb216169a15be718a35d1f6247c7b8@10.0.5.191 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_501-UA/3.1.6.0017 Accept-Language: fr-fr,fr;q=0.9,en;q=0.8 Max-Forwards: 70 Content-Length: 0 <-------------> [Jul 6 15:52:22] DEBUG[2782] chan_sip.c: Header 0 [ 30]: ACK sip:33i@10.0.5.191 SIP/2.0 [Jul 6 15:52:22] DEBUG[2782] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 10.0.5.198;branch=z9hG4bK5f206ac8AF82B9D3 [Jul 6 15:52:22] DEBUG[2782] chan_sip.c: Header 2 [ 59]: From: ;tag=5B8DFB2F-A3F8FFB4 [Jul 6 15:52:22] DEBUG[2782] chan_sip.c: Header 3 [ 45]: To: "33i" ;tag=as5a66f1a4 [Jul 6 15:52:22] DEBUG[2782] chan_sip.c: Header 4 [ 11]: CSeq: 1 ACK [Jul 6 15:52:22] DEBUG[2782] chan_sip.c: Header 5 [ 52]: Call-ID: 4bdb216169a15be718a35d1f6247c7b8@10.0.5.191 [Jul 6 15:52:22] DEBUG[2782] chan_sip.c: Header 6 [ 40]: Contact: [Jul 6 15:52:22] DEBUG[2782] chan_sip.c: Header 7 [ 96]: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER [Jul 6 15:52:22] DEBUG[2782] chan_sip.c: Header 8 [ 54]: User-Agent: PolycomSoundPointIP-SPIP_501-UA/3.1.6.0017 [Jul 6 15:52:22] DEBUG[2782] chan_sip.c: Header 9 [ 40]: Accept-Language: fr-fr,fr;q=0.9,en;q=0.8 [Jul 6 15:52:22] DEBUG[2782] chan_sip.c: Header 10 [ 16]: Max-Forwards: 70 [Jul 6 15:52:22] DEBUG[2782] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Jul 6 15:52:22] DEBUG[2782] chan_sip.c: Header 12 [ 0]: [Jul 6 15:52:22] VERBOSE[2782] chan_sip.c: --- (12 headers 0 lines) --- [Jul 6 15:52:22] DEBUG[2782] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [Jul 6 15:52:22] DEBUG[2782] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #66 [Jul 6 15:52:22] DEBUG[2782] chan_sip.c: Stopping retransmission on '4bdb216169a15be718a35d1f6247c7b8@10.0.5.191' of Response 1: Match Found [Jul 6 15:52:24] DEBUG[2907] rtp.c: Got RTCP report of 64 bytes [Jul 6 15:52:25] VERBOSE[2782] chan_sip.c: <--- SIP read from UDP:10.0.5.198:5060 ---> REFER sip:33i@10.0.5.191 SIP/2.0 Via: SIP/2.0/UDP 10.0.5.198;branch=z9hG4bK8896d32dA374B79C From: ;tag=5B8DFB2F-A3F8FFB4 To: "33i" ;tag=as5a66f1a4 CSeq: 2 REFER Call-ID: 4bdb216169a15be718a35d1f6247c7b8@10.0.5.191 Contact: User-Agent: PolycomSoundPointIP-SPIP_501-UA/3.1.6.0017 Accept-Language: fr-fr,fr;q=0.9,en;q=0.8 Refer-To: sip:10@10.0.5.191;user=phone Referred-By: Max-Forwards: 70 Content-Length: 0 <-------------> [Jul 6 15:52:25] DEBUG[2782] chan_sip.c: Header 0 [ 32]: REFER sip:33i@10.0.5.191 SIP/2.0 [Jul 6 15:52:25] DEBUG[2782] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 10.0.5.198;branch=z9hG4bK8896d32dA374B79C [Jul 6 15:52:25] DEBUG[2782] chan_sip.c: Header 2 [ 59]: From: ;tag=5B8DFB2F-A3F8FFB4 [Jul 6 15:52:25] DEBUG[2782] chan_sip.c: Header 3 [ 45]: To: "33i" ;tag=as5a66f1a4 [Jul 6 15:52:25] DEBUG[2782] chan_sip.c: Header 4 [ 13]: CSeq: 2 REFER [Jul 6 15:52:25] DEBUG[2782] chan_sip.c: Header 5 [ 52]: Call-ID: 4bdb216169a15be718a35d1f6247c7b8@10.0.5.191 [Jul 6 15:52:25] DEBUG[2782] chan_sip.c: Header 6 [ 40]: Contact: [Jul 6 15:52:25] DEBUG[2782] chan_sip.c: Header 7 [ 54]: User-Agent: PolycomSoundPointIP-SPIP_501-UA/3.1.6.0017 [Jul 6 15:52:25] DEBUG[2782] chan_sip.c: Header 8 [ 40]: Accept-Language: fr-fr,fr;q=0.9,en;q=0.8 [Jul 6 15:52:25] DEBUG[2782] chan_sip.c: Header 9 [ 38]: Refer-To: sip:10@10.0.5.191;user=phone [Jul 6 15:52:25] DEBUG[2782] chan_sip.c: Header 10 [ 44]: Referred-By: [Jul 6 15:52:25] DEBUG[2782] chan_sip.c: Header 11 [ 16]: Max-Forwards: 70 [Jul 6 15:52:25] DEBUG[2782] chan_sip.c: Header 12 [ 17]: Content-Length: 0 [Jul 6 15:52:25] DEBUG[2782] chan_sip.c: Header 13 [ 0]: [Jul 6 15:52:25] VERBOSE[2782] chan_sip.c: --- (13 headers 0 lines) --- [Jul 6 15:52:25] DEBUG[2782] chan_sip.c: **** Received REFER (9) - Command in SIP REFER [Jul 6 15:52:25] VERBOSE[2782] chan_sip.c: Call 4bdb216169a15be718a35d1f6247c7b8@10.0.5.191 got a SIP call transfer from caller: (REFER)! [Jul 6 15:52:25] VERBOSE[2782] chan_sip.c: SIP transfer to extension 10@from-sip by 0004f215aabb-1@10.0.5.191 [Jul 6 15:52:25] DEBUG[2782] chan_sip.c: SIP blind transfer: Transferer channel SIP/0004f215aabb-1-00000001, transferee channel SIP/33i-00000000 [Jul 6 15:52:25] DEBUG[2782] chan_sip.c: Got SIP transfer, applying to bridged peer 'SIP/33i-00000000' [Jul 6 15:52:25] VERBOSE[2782] chan_sip.c: <--- Transmitting (no NAT) to 10.0.5.198:5060 ---> SIP/2.0 202 Accepted Via: SIP/2.0/UDP 10.0.5.198;branch=z9hG4bK8896d32dA374B79C;received=10.0.5.198 From: ;tag=5B8DFB2F-A3F8FFB4 To: "33i" ;tag=as5a66f1a4 Call-ID: 4bdb216169a15be718a35d1f6247c7b8@10.0.5.191 CSeq: 2 REFER Server: Asterisk PBX 1.6.2.9 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: Content-Length: 0 <------------> [Jul 6 15:52:25] DEBUG[2782] chan_sip.c: Trying to put 'SIP/2.0 202' onto UDP socket destined for 10.0.5.198:5060 [Jul 6 15:52:25] DEBUG[2782] chan_sip.c: chan1->name: SIP/0004f215aabb-1-00000001 [Jul 6 15:52:25] DEBUG[2782] chan_sip.c: Strict routing enforced for session 4bdb216169a15be718a35d1f6247c7b8@10.0.5.191 [Jul 6 15:52:25] VERBOSE[2782] chan_sip.c: set_destination: Parsing for address/port to send to [Jul 6 15:52:25] VERBOSE[2782] chan_sip.c: set_destination: set destination to 10.0.5.198, port 5060 [Jul 6 15:52:25] VERBOSE[2782] chan_sip.c: Reliably Transmitting (no NAT) to 10.0.5.198:5060: NOTIFY sip:0004f215aabb-1@10.0.5.198 SIP/2.0 Via: SIP/2.0/UDP 10.0.5.191:5060;branch=z9hG4bK25efecd8;rport Max-Forwards: 70 From: "33i" ;tag=as5a66f1a4 To: ;tag=5B8DFB2F-A3F8FFB4 Contact: Call-ID: 4bdb216169a15be718a35d1f6247c7b8@10.0.5.191 CSeq: 104 NOTIFY User-Agent: Asterisk PBX 1.6.2.9 Event: refer;id=2 Subscription-state: active Content-Type: message/sipfrag;version=2.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 21 SIP/2.0 183 Ringing --- [Jul 6 15:52:25] DEBUG[2782] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #68 [Jul 6 15:52:25] DEBUG[2782] chan_sip.c: Trying to put 'NOTIFY sip:' onto UDP socket destined for 10.0.5.198:5060 [Jul 6 15:52:25] DEBUG[2782] channel.c: Soft-Hanging up channel 'SIP/33i-00000000' [Jul 6 15:52:25] DEBUG[2782] chan_sip.c: Blind transfer succeeded. Telling transferer. [Jul 6 15:52:25] DEBUG[2782] chan_sip.c: Strict routing enforced for session 4bdb216169a15be718a35d1f6247c7b8@10.0.5.191 [Jul 6 15:52:25] VERBOSE[2782] chan_sip.c: set_destination: Parsing for address/port to send to [Jul 6 15:52:25] VERBOSE[2782] chan_sip.c: set_destination: set destination to 10.0.5.198, port 5060 [Jul 6 15:52:25] VERBOSE[2782] chan_sip.c: Reliably Transmitting (no NAT) to 10.0.5.198:5060: NOTIFY sip:0004f215aabb-1@10.0.5.198 SIP/2.0 Via: SIP/2.0/UDP 10.0.5.191:5060;branch=z9hG4bK271e4114;rport Max-Forwards: 70 From: "33i" ;tag=as5a66f1a4 To: ;tag=5B8DFB2F-A3F8FFB4 Contact: Call-ID: 4bdb216169a15be718a35d1f6247c7b8@10.0.5.191 CSeq: 105 NOTIFY User-Agent: Asterisk PBX 1.6.2.9 Event: refer;id=2 Subscription-state: terminated;reason=noresource Content-Type: message/sipfrag;version=2.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 16 SIP/2.0 200 Ok --- [Jul 6 15:52:25] DEBUG[2782] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #69 [Jul 6 15:52:25] DEBUG[2782] chan_sip.c: Trying to put 'NOTIFY sip:' onto UDP socket destined for 10.0.5.198:5060 [Jul 6 15:52:25] VERBOSE[2907] res_musiconhold.c: -- Stopped music on hold on SIP/33i-00000000 [Jul 6 15:52:25] DEBUG[2907] channel.c: Set channel SIP/33i-00000000 to write format ulaw [Jul 6 15:52:25] DEBUG[2907] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Jul 6 15:52:25] DEBUG[2907] rtp.c: Oooh, got a hangup [Jul 6 15:52:25] DEBUG[2907] channel.c: Returning from native bridge, channels: SIP/33i-00000000, SIP/0004f215aabb-1-00000001 [Jul 6 15:52:25] DEBUG[2907] channel.c: Hanging up channel 'SIP/0004f215aabb-1-00000001' [Jul 6 15:52:25] DEBUG[2907] chan_sip.c: update_call_counter(test) - decrement call limit counter on hangup [Jul 6 15:52:25] DEBUG[2907] chan_sip.c: Updating call counter for outgoing call [Jul 6 15:52:25] DEBUG[2907] chan_sip.c: Call to peer '0004f215aabb-1' removed from call limit 0 [Jul 6 15:52:25] DEBUG[2907] chan_sip.c: SIP Transfer: Not hanging up right now... Rescheduling hangup for 4bdb216169a15be718a35d1f6247c7b8@10.0.5.191. [Jul 6 15:52:25] VERBOSE[2907] chan_sip.c: Scheduling destruction of SIP dialog '4bdb216169a15be718a35d1f6247c7b8@10.0.5.191' in 32000 ms (Method: REFER) [Jul 6 15:52:25] DEBUG[2907] rtp.c: Channel '' has no RTP, not doing anything [Jul 6 15:52:25] DEBUG[2907] app_dial.c: Exiting with DIALSTATUS=ANSWER. [Jul 6 15:52:25] DEBUG[2907] pbx.c: Spawn extension (from-sip,10,1) exited non-zero on 'SIP/33i-00000000' [Jul 6 15:52:25] VERBOSE[2907] pbx.c: == Spawn extension (from-sip, 10, 1) exited non-zero on 'SIP/33i-00000000' [Jul 6 15:52:25] DEBUG[2907] pbx.c: Launching 'NoOp' [Jul 6 15:52:25] VERBOSE[2907] pbx.c: -- Executing [10@from-sip:1] NoOp("SIP/33i-00000000", "") in new stack [Jul 6 15:52:25] DEBUG[2907] pbx.c: Launching 'Park' [Jul 6 15:52:25] VERBOSE[2907] pbx.c: -- Executing [10@from-sip:2] Park("SIP/33i-00000000", "") in new stack [Jul 6 15:52:25] DEBUG[2689] devicestate.c: No provider found, checking channel drivers for SIP - 0004f215aabb-1 [Jul 6 15:52:25] DEBUG[2689] chan_sip.c: Checking device state for peer 0004f215aabb-1 [Jul 6 15:52:25] DEBUG[2689] devicestate.c: Changing state for SIP/0004f215aabb-1 - state 1 (Not in use) [Jul 6 15:52:25] DEBUG[2689] devicestate.c: device 'SIP/0004f215aabb-1' state '1' [Jul 6 15:52:25] DEBUG[2689] devicestate.c: No provider found, checking channel drivers for SIP - 0004f215aabb-1 [Jul 6 15:52:25] DEBUG[2689] chan_sip.c: Checking device state for peer 0004f215aabb-1 [Jul 6 15:52:25] DEBUG[2689] devicestate.c: Changing state for SIP/0004f215aabb-1 - state 1 (Not in use) [Jul 6 15:52:25] DEBUG[2689] devicestate.c: device 'SIP/0004f215aabb-1' state '1' [Jul 6 15:52:25] DEBUG[2779] app_queue.c: Device 'SIP/0004f215aabb-1' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Jul 6 15:52:25] DEBUG[2779] app_queue.c: Device 'SIP/0004f215aabb-1' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Jul 6 15:52:25] VERBOSE[2782] chan_sip.c: <--- SIP read from UDP:10.0.5.198:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.5.191:5060;branch=z9hG4bK25efecd8;rport From: "33i" ;tag=as5a66f1a4 To: ;tag=5B8DFB2F-A3F8FFB4 CSeq: 104 NOTIFY Call-ID: 4bdb216169a15be718a35d1f6247c7b8@10.0.5.191 Contact: Event: refer;id=2 User-Agent: PolycomSoundPointIP-SPIP_501-UA/3.1.6.0017 Accept-Language: fr-fr,fr;q=0.9,en;q=0.8 Content-Length: 0 <-------------> [Jul 6 15:52:25] DEBUG[2782] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Jul 6 15:52:25] DEBUG[2782] chan_sip.c: Header 1 [ 61]: Via: SIP/2.0/UDP 10.0.5.191:5060;branch=z9hG4bK25efecd8;rport [Jul 6 15:52:25] DEBUG[2782] chan_sip.c: Header 2 [ 47]: From: "33i" ;tag=as5a66f1a4 [Jul 6 15:52:25] DEBUG[2782] chan_sip.c: Header 3 [ 57]: To: ;tag=5B8DFB2F-A3F8FFB4 [Jul 6 15:52:25] DEBUG[2782] chan_sip.c: Header 4 [ 16]: CSeq: 104 NOTIFY [Jul 6 15:52:25] DEBUG[2782] chan_sip.c: Header 5 [ 52]: Call-ID: 4bdb216169a15be718a35d1f6247c7b8@10.0.5.191 [Jul 6 15:52:25] DEBUG[2782] chan_sip.c: Header 6 [ 40]: Contact: [Jul 6 15:52:25] DEBUG[2782] chan_sip.c: Header 7 [ 17]: Event: refer;id=2 [Jul 6 15:52:25] DEBUG[2782] chan_sip.c: Header 8 [ 54]: User-Agent: PolycomSoundPointIP-SPIP_501-UA/3.1.6.0017 [Jul 6 15:52:25] DEBUG[2782] chan_sip.c: Header 9 [ 40]: Accept-Language: fr-fr,fr;q=0.9,en;q=0.8 [Jul 6 15:52:25] DEBUG[2782] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Jul 6 15:52:25] DEBUG[2782] chan_sip.c: Header 11 [ 0]: [Jul 6 15:52:25] VERBOSE[2782] chan_sip.c: --- (11 headers 0 lines) --- [Jul 6 15:52:25] DEBUG[2782] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #68 [Jul 6 15:52:25] DEBUG[2782] chan_sip.c: Stopping retransmission on '4bdb216169a15be718a35d1f6247c7b8@10.0.5.191' of Request 104: Match Found [Jul 6 15:52:25] VERBOSE[2782] chan_sip.c: SIP Response message for INCOMING dialog NOTIFY arrived [Jul 6 15:52:25] VERBOSE[2782] chan_sip.c: <--- SIP read from UDP:10.0.5.198:5060 ---> BYE sip:33i@10.0.5.191 SIP/2.0 Via: SIP/2.0/UDP 10.0.5.198;branch=z9hG4bKae2d1d37266B056E From: ;tag=5B8DFB2F-A3F8FFB4 To: "33i" ;tag=as5a66f1a4 CSeq: 3 BYE Call-ID: 4bdb216169a15be718a35d1f6247c7b8@10.0.5.191 Contact: User-Agent: PolycomSoundPointIP-SPIP_501-UA/3.1.6.0017 Accept-Language: fr-fr,fr;q=0.9,en;q=0.8 Max-Forwards: 70 Content-Length: 0 <-------------> [Jul 6 15:52:25] DEBUG[2782] chan_sip.c: Header 0 [ 30]: BYE sip:33i@10.0.5.191 SIP/2.0 [Jul 6 15:52:25] DEBUG[2782] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 10.0.5.198;branch=z9hG4bKae2d1d37266B056E [Jul 6 15:52:25] DEBUG[2782] chan_sip.c: Header 2 [ 59]: From: ;tag=5B8DFB2F-A3F8FFB4 [Jul 6 15:52:25] DEBUG[2782] chan_sip.c: Header 3 [ 45]: To: "33i" ;tag=as5a66f1a4 [Jul 6 15:52:25] DEBUG[2782] chan_sip.c: Header 4 [ 11]: CSeq: 3 BYE [Jul 6 15:52:25] DEBUG[2782] chan_sip.c: Header 5 [ 52]: Call-ID: 4bdb216169a15be718a35d1f6247c7b8@10.0.5.191 [Jul 6 15:52:25] DEBUG[2782] chan_sip.c: Header 6 [ 40]: Contact: [Jul 6 15:52:25] DEBUG[2782] chan_sip.c: Header 7 [ 54]: User-Agent: PolycomSoundPointIP-SPIP_501-UA/3.1.6.0017 [Jul 6 15:52:25] DEBUG[2782] chan_sip.c: Header 8 [ 40]: Accept-Language: fr-fr,fr;q=0.9,en;q=0.8 [Jul 6 15:52:25] DEBUG[2782] chan_sip.c: Header 9 [ 16]: Max-Forwards: 70 [Jul 6 15:52:25] DEBUG[2782] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Jul 6 15:52:25] DEBUG[2782] chan_sip.c: Header 11 [ 0]: [Jul 6 15:52:25] VERBOSE[2782] chan_sip.c: --- (11 headers 0 lines) --- [Jul 6 15:52:25] DEBUG[2782] chan_sip.c: **** Received BYE (8) - Command in SIP BYE [Jul 6 15:52:25] DEBUG[2782] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #69 [Jul 6 15:52:25] DEBUG[2782] chan_sip.c: Stopping retransmission on '4bdb216169a15be718a35d1f6247c7b8@10.0.5.191' of Request 105: Match Found [Jul 6 15:52:25] DEBUG[2782] chan_sip.c: Initializing initreq for method BYE - callid 4bdb216169a15be718a35d1f6247c7b8@10.0.5.191 [Jul 6 15:52:25] VERBOSE[2782] chan_sip.c: Sending to 10.0.5.198 : 5060 (no NAT) [Jul 6 15:52:25] DEBUG[2782] chan_sip.c: Setting SIP_ALREADYGONE on dialog 4bdb216169a15be718a35d1f6247c7b8@10.0.5.191 [Jul 6 15:52:25] VERBOSE[2782] chan_sip.c: Scheduling destruction of SIP dialog '4bdb216169a15be718a35d1f6247c7b8@10.0.5.191' in 32000 ms (Method: BYE) [Jul 6 15:52:25] DEBUG[2782] chan_sip.c: Received bye, no owner, selfdestruct soon. [Jul 6 15:52:25] VERBOSE[2782] chan_sip.c: <--- Transmitting (no NAT) to 10.0.5.198:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.5.198;branch=z9hG4bKae2d1d37266B056E;received=10.0.5.198 From: ;tag=5B8DFB2F-A3F8FFB4 To: "33i" ;tag=as5a66f1a4 Call-ID: 4bdb216169a15be718a35d1f6247c7b8@10.0.5.191 CSeq: 3 BYE Server: Asterisk PBX 1.6.2.9 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 <------------> [Jul 6 15:52:25] DEBUG[2782] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.0.5.198:5060 [Jul 6 15:52:25] VERBOSE[2782] chan_sip.c: <--- SIP read from UDP:10.0.5.198:5060 ---> SIP/2.0 481 Call Leg/Transaction Does Not Exist Via: SIP/2.0/UDP 10.0.5.191:5060;branch=z9hG4bK271e4114;rport From: "33i" ;tag=as5a66f1a4 To: ;tag=5B8DFB2F-A3F8FFB4 CSeq: 105 NOTIFY Call-ID: 4bdb216169a15be718a35d1f6247c7b8@10.0.5.191 Event: refer;id=2 User-Agent: PolycomSoundPointIP-SPIP_501-UA/3.1.6.0017 Accept-Language: fr-fr,fr;q=0.9,en;q=0.8 Content-Length: 0 <-------------> [Jul 6 15:52:25] DEBUG[2782] chan_sip.c: Header 0 [ 47]: SIP/2.0 481 Call Leg/Transaction Does Not Exist [Jul 6 15:52:25] DEBUG[2782] chan_sip.c: Header 1 [ 61]: Via: SIP/2.0/UDP 10.0.5.191:5060;branch=z9hG4bK271e4114;rport [Jul 6 15:52:25] DEBUG[2782] chan_sip.c: Header 2 [ 47]: From: "33i" ;tag=as5a66f1a4 [Jul 6 15:52:25] DEBUG[2782] chan_sip.c: Header 3 [ 57]: To: ;tag=5B8DFB2F-A3F8FFB4 [Jul 6 15:52:25] DEBUG[2782] chan_sip.c: Header 4 [ 16]: CSeq: 105 NOTIFY [Jul 6 15:52:25] DEBUG[2782] chan_sip.c: Header 5 [ 52]: Call-ID: 4bdb216169a15be718a35d1f6247c7b8@10.0.5.191 [Jul 6 15:52:25] DEBUG[2782] chan_sip.c: Header 6 [ 17]: Event: refer;id=2 [Jul 6 15:52:25] DEBUG[2782] chan_sip.c: Header 7 [ 54]: User-Agent: PolycomSoundPointIP-SPIP_501-UA/3.1.6.0017 [Jul 6 15:52:25] DEBUG[2782] chan_sip.c: Header 8 [ 40]: Accept-Language: fr-fr,fr;q=0.9,en;q=0.8 [Jul 6 15:52:25] DEBUG[2782] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Jul 6 15:52:25] DEBUG[2782] chan_sip.c: Header 10 [ 0]: [Jul 6 15:52:25] VERBOSE[2782] chan_sip.c: --- (10 headers 0 lines) --- [Jul 6 15:52:25] DEBUG[2782] chan_sip.c: Stopping retransmission on '4bdb216169a15be718a35d1f6247c7b8@10.0.5.191' of Request 105: Match Not Found [Jul 6 15:52:26] DEBUG[2907] features.c: Multiparking: default refcount now 2 [Jul 6 15:52:26] DEBUG[2907] features.c: Parkinglot: default [Jul 6 15:52:26] DEBUG[2907] features.c: Multiparking: default refcount now 3 [Jul 6 15:52:26] DEBUG[2907] features.c: Multiparking: default refcount now 2 [Jul 6 15:52:26] DEBUG[2907] channel.c: Planning to masquerade channel SIP/33i-00000000 into the structure of Parked/SIP/33i-00000000 [Jul 6 15:52:26] DEBUG[2907] channel.c: Done planning to masquerade channel SIP/33i-00000000 into the structure of Parked/SIP/33i-00000000 [Jul 6 15:52:26] DEBUG[2907] channel.c: Actually Masquerading SIP/33i-00000000(6) into the structure of Parked/SIP/33i-00000000(0) [Jul 6 15:52:26] DEBUG[2907] channel.c: Got clone lock for masquerade on 'SIP/33i-00000000' at 0x878f788 [Jul 6 15:52:26] DEBUG[2907] channel.c: Putting channel SIP/33i-00000000 in 4/4 formats [Jul 6 15:52:26] DEBUG[2907] chan_sip.c: SIP Fixup: New owner for dialogue f7bf38449b305573: SIP/33i-00000000 (Old parent: Parked/SIP/33i-00000000) [Jul 6 15:52:26] DEBUG[2907] channel.c: Released clone lock on 'Parked/SIP/33i-00000000' [Jul 6 15:52:26] DEBUG[2907] channel.c: Done Masquerading SIP/33i-00000000 (6) [Jul 6 15:52:26] VERBOSE[2907] features.c: == Parked SIP/33i-00000000 on 701 (lot default). Will timeout back to extension [from-sip] s, 1 in 45 seconds [Jul 6 15:52:26] DEBUG[2907] pbx.c: Added extension '701' priority 1 to parkedcalls (0x87688f0) [Jul 6 15:52:26] VERBOSE[2907] pbx.c: -- Added extension '701' priority 1 to parkedcalls (0x87688f0) [Jul 6 15:52:26] DEBUG[2907] features.c: Notification of state change to metermaids 701@parkedcalls to state 'In use'[Jul 6 15:52:26] DEBUG[2907] devicestate.c: device 'park:701@parkedcalls' state '2' [Jul 6 15:52:26] DEBUG[2690] chan_sip.c: Strict routing enforced for session ab3922f0-ae400522-a64ac11b@10.0.5.198 [Jul 6 15:52:26] VERBOSE[2690] chan_sip.c: set_destination: Parsing for address/port to send to [Jul 6 15:52:26] VERBOSE[2690] chan_sip.c: set_destination: set destination to 10.0.5.198, port 5060 [Jul 6 15:52:26] VERBOSE[2690] chan_sip.c: Reliably Transmitting (no NAT) to 10.0.5.198:5060: NOTIFY sip:0004f215aabb-1@10.0.5.198 SIP/2.0 Via: SIP/2.0/UDP 10.0.5.191:5060;branch=z9hG4bK256a2d45;rport Max-Forwards: 70 From: ;tag=as545536d7 To: "0004f215aabb-1" ;tag=B04C9344-7C4E7635 Contact: Call-ID: ab3922f0-ae400522-a64ac11b@10.0.5.198 CSeq: 103 NOTIFY User-Agent: Asterisk PBX 1.6.2.9 Event: presence Content-Type: application/xpidf+xml Subscription-State: active Content-Length: 353
--- [Jul 6 15:52:26] DEBUG[2690] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #72 [Jul 6 15:52:26] DEBUG[2690] chan_sip.c: Trying to put 'NOTIFY sip:' onto UDP socket destined for 10.0.5.198:5060 [Jul 6 15:52:26] VERBOSE[2690] chan_sip.c: == Extension Changed 800[5555555555-hints] new state InUse for Notify User test [Jul 6 15:52:26] DEBUG[2779] app_queue.c: Device 'park:701@parkedcalls' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Jul 6 15:52:26] DEBUG[2907] channel.c: Set channel SIP/33i-00000000 to write format gsm [Jul 6 15:52:26] DEBUG[2907] rtp.c: Difference is 8744, ms is 1113 [Jul 6 15:52:26] DEBUG[2907] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [Jul 6 15:52:26] VERBOSE[2907] file.c: -- Playing 'digits/7.gsm' (language 'en') [Jul 6 15:52:26] VERBOSE[2782] chan_sip.c: <--- SIP read from UDP:10.0.5.198:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.5.191:5060;branch=z9hG4bK256a2d45;rport From: ;tag=as545536d7 To: "0004f215aabb-1" ;tag=B04C9344-7C4E7635 CSeq: 103 NOTIFY Call-ID: ab3922f0-ae400522-a64ac11b@10.0.5.198 Contact: Event: presence User-Agent: PolycomSoundPointIP-SPIP_501-UA/3.1.6.0017 Accept-Language: fr-fr,fr;q=0.9,en;q=0.8 Content-Length: 0 <-------------> [Jul 6 15:52:26] DEBUG[2782] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Jul 6 15:52:26] DEBUG[2782] chan_sip.c: Header 1 [ 61]: Via: SIP/2.0/UDP 10.0.5.191:5060;branch=z9hG4bK256a2d45;rport [Jul 6 15:52:26] DEBUG[2782] chan_sip.c: Header 2 [ 41]: From: ;tag=as545536d7 [Jul 6 15:52:26] DEBUG[2782] chan_sip.c: Header 3 [ 74]: To: "0004f215aabb-1" ;tag=B04C9344-7C4E7635 [Jul 6 15:52:26] DEBUG[2782] chan_sip.c: Header 4 [ 16]: CSeq: 103 NOTIFY [Jul 6 15:52:26] DEBUG[2782] chan_sip.c: Header 5 [ 46]: Call-ID: ab3922f0-ae400522-a64ac11b@10.0.5.198 [Jul 6 15:52:26] DEBUG[2782] chan_sip.c: Header 6 [ 40]: Contact: [Jul 6 15:52:26] DEBUG[2782] chan_sip.c: Header 7 [ 15]: Event: presence [Jul 6 15:52:26] DEBUG[2782] chan_sip.c: Header 8 [ 54]: User-Agent: PolycomSoundPointIP-SPIP_501-UA/3.1.6.0017 [Jul 6 15:52:26] DEBUG[2782] chan_sip.c: Header 9 [ 40]: Accept-Language: fr-fr,fr;q=0.9,en;q=0.8 [Jul 6 15:52:26] DEBUG[2782] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Jul 6 15:52:26] DEBUG[2782] chan_sip.c: Header 11 [ 0]: [Jul 6 15:52:26] VERBOSE[2782] chan_sip.c: --- (11 headers 0 lines) --- [Jul 6 15:52:26] DEBUG[2782] chan_sip.c: Acked pending invite 103 [Jul 6 15:52:26] DEBUG[2782] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #72 [Jul 6 15:52:26] DEBUG[2782] chan_sip.c: Stopping retransmission on 'ab3922f0-ae400522-a64ac11b@10.0.5.198' of Request 103: Match Found [Jul 6 15:52:26] VERBOSE[2782] chan_sip.c: SIP Response message for INCOMING dialog NOTIFY arrived [Jul 6 15:52:27] DEBUG[2907] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Jul 6 15:52:27] DEBUG[2907] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Jul 6 15:52:27] DEBUG[2907] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Jul 6 15:52:27] DEBUG[2907] channel.c: Set channel SIP/33i-00000000 to write format ulaw [Jul 6 15:52:27] DEBUG[2907] channel.c: Set channel SIP/33i-00000000 to write format gsm [Jul 6 15:52:27] DEBUG[2907] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [Jul 6 15:52:27] VERBOSE[2907] file.c: -- Playing 'digits/0.gsm' (language 'en') [Jul 6 15:52:27] NOTICE[2907] rtp.c: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: 10.0.5.196 [Jul 6 15:52:27] DEBUG[2907] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Jul 6 15:52:27] DEBUG[2907] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Jul 6 15:52:27] DEBUG[2907] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Jul 6 15:52:27] DEBUG[2907] channel.c: Set channel SIP/33i-00000000 to write format ulaw [Jul 6 15:52:27] DEBUG[2907] channel.c: Set channel SIP/33i-00000000 to write format gsm [Jul 6 15:52:27] DEBUG[2907] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [Jul 6 15:52:27] VERBOSE[2907] file.c: -- Playing 'digits/1.gsm' (language 'en') [Jul 6 15:52:28] DEBUG[2907] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Jul 6 15:52:28] DEBUG[2907] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Jul 6 15:52:28] DEBUG[2907] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Jul 6 15:52:28] DEBUG[2907] channel.c: Set channel SIP/33i-00000000 to write format ulaw [Jul 6 15:52:28] DEBUG[2907] rtp.c: Setting the marker bit due to a source update [Jul 6 15:52:28] VERBOSE[2907] res_musiconhold.c: -- Started music on hold, class 'default', on SIP/33i-00000000 [Jul 6 15:52:28] DEBUG[2907] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [Jul 6 15:52:28] DEBUG[2907] pbx.c: Spawn extension (from-sip,s,1) exited non-zero on 'Parked/SIP/33i-00000000' [Jul 6 15:52:28] VERBOSE[2907] pbx.c: == Spawn extension (from-sip, s, 1) exited non-zero on 'Parked/SIP/33i-00000000' [Jul 6 15:52:28] DEBUG[2907] channel.c: Soft-Hanging up channel 'Parked/SIP/33i-00000000' [Jul 6 15:52:28] DEBUG[2907] channel.c: Hanging up zombie 'Parked/SIP/33i-00000000' [Jul 6 15:52:28] DEBUG[2689] devicestate.c: No provider found, checking channel drivers for Parked - SIP/33i [Jul 6 15:52:28] DEBUG[2689] devicestate.c: Changing state for Parked/SIP/33i - state 4 (Invalid) [Jul 6 15:52:28] DEBUG[2689] devicestate.c: device 'Parked/SIP/33i' state '4' [Jul 6 15:52:28] DEBUG[2779] app_queue.c: Device 'Parked/SIP/33i' changed to state '4' (Invalid) but we don't care because they're not a member of any queue. [Jul 6 15:52:28] DEBUG[2695] channel.c: Set channel SIP/33i-00000000 to write format slin [Jul 6 15:52:28] DEBUG[2695] res_musiconhold.c: SIP/33i-00000000 Opened file 1 '/var/lib/asterisk/moh/manolo_camp-morning_coffee' [Jul 6 15:52:29] DEBUG[2695] rtp.c: Got RTCP report of 64 bytes [Jul 6 15:52:34] DEBUG[2695] rtp.c: Got RTCP report of 64 bytes [Jul 6 15:52:35] VERBOSE[2782] chan_sip.c: <--- SIP read from UDP:10.0.5.198:5060 ---> REGISTER sip:10.0.5.191 SIP/2.0 Via: SIP/2.0/UDP 10.0.5.198;branch=z9hG4bKddf2d6715C3DA30 From: "0004f215aabb-1" ;tag=6F5D75EC-D185443D To: CSeq: 11 REGISTER Call-ID: ad2a1218-82f9b90a-393cb63@10.0.5.198 Contact: ;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER" User-Agent: PolycomSoundPointIP-SPIP_501-UA/3.1.6.0017 Accept-Language: fr-fr,fr;q=0.9,en;q=0.8 Authorization: Digest username="0004f215aabb-1", realm="asterisk", nonce="33987b61", uri="sip:10.0.5.191", response="6e917c45f50f454054781e5ca18413e0", algorithm=MD5 Max-Forwards: 70 Expires: 60 Content-Length: 0 <-------------> [Jul 6 15:52:35] DEBUG[2782] chan_sip.c: Header 0 [ 31]: REGISTER sip:10.0.5.191 SIP/2.0 [Jul 6 15:52:35] DEBUG[2782] chan_sip.c: Header 1 [ 57]: Via: SIP/2.0/UDP 10.0.5.198;branch=z9hG4bKddf2d6715C3DA30 [Jul 6 15:52:35] DEBUG[2782] chan_sip.c: Header 2 [ 76]: From: "0004f215aabb-1" ;tag=6F5D75EC-D185443D [Jul 6 15:52:35] DEBUG[2782] chan_sip.c: Header 3 [ 35]: To: [Jul 6 15:52:35] DEBUG[2782] chan_sip.c: Header 4 [ 17]: CSeq: 11 REGISTER [Jul 6 15:52:35] DEBUG[2782] chan_sip.c: Header 5 [ 45]: Call-ID: ad2a1218-82f9b90a-393cb63@10.0.5.198 [Jul 6 15:52:35] DEBUG[2782] chan_sip.c: Header 6 [140]: Contact: ;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER" [Jul 6 15:52:35] DEBUG[2782] chan_sip.c: Header 7 [ 54]: User-Agent: PolycomSoundPointIP-SPIP_501-UA/3.1.6.0017 [Jul 6 15:52:35] DEBUG[2782] chan_sip.c: Header 8 [ 40]: Accept-Language: fr-fr,fr;q=0.9,en;q=0.8 [Jul 6 15:52:35] DEBUG[2782] chan_sip.c: Header 9 [165]: Authorization: Digest username="0004f215aabb-1", realm="asterisk", nonce="33987b61", uri="sip:10.0.5.191", response="6e917c45f50f454054781e5ca18413e0", algorithm=MD5 [Jul 6 15:52:35] DEBUG[2782] chan_sip.c: Header 10 [ 16]: Max-Forwards: 70 [Jul 6 15:52:35] DEBUG[2782] chan_sip.c: Header 11 [ 11]: Expires: 60 [Jul 6 15:52:35] DEBUG[2782] chan_sip.c: Header 12 [ 17]: Content-Length: 0 [Jul 6 15:52:35] DEBUG[2782] chan_sip.c: Header 13 [ 0]: [Jul 6 15:52:35] VERBOSE[2782] chan_sip.c: --- (13 headers 0 lines) --- [Jul 6 15:52:35] DEBUG[2782] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Jul 6 15:52:35] DEBUG[2782] chan_sip.c: Initializing initreq for method REGISTER - callid ad2a1218-82f9b90a-393cb63@10.0.5.198 [Jul 6 15:52:35] VERBOSE[2782] chan_sip.c: Sending to 10.0.5.198 : 5060 (no NAT) [Jul 6 15:52:35] NOTICE[2782] chan_sip.c: Correct auth, but based on stale nonce received from '' [Jul 6 15:52:35] VERBOSE[2782] chan_sip.c: <--- Transmitting (no NAT) to 10.0.5.198:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.0.5.198;branch=z9hG4bKddf2d6715C3DA30;received=10.0.5.198 From: "0004f215aabb-1" ;tag=6F5D75EC-D185443D To: ;tag=as543567ce Call-ID: ad2a1218-82f9b90a-393cb63@10.0.5.198 CSeq: 11 REGISTER Server: Asterisk PBX 1.6.2.9 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1961ef9a", stale=true Content-Length: 0 <------------> [Jul 6 15:52:35] DEBUG[2782] chan_sip.c: Trying to put 'SIP/2.0 401' onto UDP socket destined for 10.0.5.198:5060 [Jul 6 15:52:35] VERBOSE[2782] chan_sip.c: Scheduling destruction of SIP dialog 'ad2a1218-82f9b90a-393cb63@10.0.5.198' in 32000 ms (Method: REGISTER) [Jul 6 15:52:35] VERBOSE[2782] chan_sip.c: <--- SIP read from UDP:10.0.5.198:5060 ---> REGISTER sip:10.0.5.191 SIP/2.0 Via: SIP/2.0/UDP 10.0.5.198;branch=z9hG4bKff8295b118D7262 From: "0004f215aabb-1" ;tag=6F5D75EC-D185443D To: CSeq: 12 REGISTER Call-ID: ad2a1218-82f9b90a-393cb63@10.0.5.198 Contact: ;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER" User-Agent: PolycomSoundPointIP-SPIP_501-UA/3.1.6.0017 Accept-Language: fr-fr,fr;q=0.9,en;q=0.8 Authorization: Digest username="0004f215aabb-1", realm="asterisk", nonce="1961ef9a", uri="sip:10.0.5.191", response="013c587edf5c4be5b37a420fb3d73ba4", algorithm=MD5 Max-Forwards: 70 Expires: 60 Content-Length: 0 <-------------> [Jul 6 15:52:35] DEBUG[2782] chan_sip.c: Header 0 [ 31]: REGISTER sip:10.0.5.191 SIP/2.0 [Jul 6 15:52:35] DEBUG[2782] chan_sip.c: Header 1 [ 57]: Via: SIP/2.0/UDP 10.0.5.198;branch=z9hG4bKff8295b118D7262 [Jul 6 15:52:35] DEBUG[2782] chan_sip.c: Header 2 [ 76]: From: "0004f215aabb-1" ;tag=6F5D75EC-D185443D [Jul 6 15:52:35] DEBUG[2782] chan_sip.c: Header 3 [ 35]: To: [Jul 6 15:52:35] DEBUG[2782] chan_sip.c: Header 4 [ 17]: CSeq: 12 REGISTER [Jul 6 15:52:35] DEBUG[2782] chan_sip.c: Header 5 [ 45]: Call-ID: ad2a1218-82f9b90a-393cb63@10.0.5.198 [Jul 6 15:52:35] DEBUG[2782] chan_sip.c: Header 6 [140]: Contact: ;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER" [Jul 6 15:52:35] DEBUG[2782] chan_sip.c: Header 7 [ 54]: User-Agent: PolycomSoundPointIP-SPIP_501-UA/3.1.6.0017 [Jul 6 15:52:35] DEBUG[2782] chan_sip.c: Header 8 [ 40]: Accept-Language: fr-fr,fr;q=0.9,en;q=0.8 [Jul 6 15:52:35] DEBUG[2782] chan_sip.c: Header 9 [165]: Authorization: Digest username="0004f215aabb-1", realm="asterisk", nonce="1961ef9a", uri="sip:10.0.5.191", response="013c587edf5c4be5b37a420fb3d73ba4", algorithm=MD5 [Jul 6 15:52:35] DEBUG[2782] chan_sip.c: Header 10 [ 16]: Max-Forwards: 70 [Jul 6 15:52:35] DEBUG[2782] chan_sip.c: Header 11 [ 11]: Expires: 60 [Jul 6 15:52:35] DEBUG[2782] chan_sip.c: Header 12 [ 17]: Content-Length: 0 [Jul 6 15:52:35] DEBUG[2782] chan_sip.c: Header 13 [ 0]: [Jul 6 15:52:35] VERBOSE[2782] chan_sip.c: --- (13 headers 0 lines) --- [Jul 6 15:52:35] DEBUG[2782] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Jul 6 15:52:35] DEBUG[2782] chan_sip.c: Initializing initreq for method REGISTER - callid ad2a1218-82f9b90a-393cb63@10.0.5.198 [Jul 6 15:52:35] VERBOSE[2782] chan_sip.c: Sending to 10.0.5.198 : 5060 (no NAT) [Jul 6 15:52:35] VERBOSE[2782] chan_sip.c: <--- Transmitting (no NAT) to 10.0.5.198:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.5.198;branch=z9hG4bKff8295b118D7262;received=10.0.5.198 From: "0004f215aabb-1" ;tag=6F5D75EC-D185443D To: ;tag=as543567ce Call-ID: ad2a1218-82f9b90a-393cb63@10.0.5.198 CSeq: 12 REGISTER Server: Asterisk PBX 1.6.2.9 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Expires: 60 Contact: ;expires=60 Date: Tue, 06 Jul 2010 19:52:35 GMT Content-Length: 0 <------------> [Jul 6 15:52:35] DEBUG[2782] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.0.5.198:5060 [Jul 6 15:52:35] VERBOSE[2782] chan_sip.c: Scheduling destruction of SIP dialog 'ad2a1218-82f9b90a-393cb63@10.0.5.198' in 32000 ms (Method: REGISTER) [Jul 6 15:52:35] DEBUG[2689] devicestate.c: No provider found, checking channel drivers for SIP - 0004f215aabb-1 [Jul 6 15:52:35] DEBUG[2689] chan_sip.c: Checking device state for peer 0004f215aabb-1 [Jul 6 15:52:35] DEBUG[2689] devicestate.c: Changing state for SIP/0004f215aabb-1 - state 1 (Not in use) [Jul 6 15:52:35] DEBUG[2689] devicestate.c: device 'SIP/0004f215aabb-1' state '1' [Jul 6 15:52:35] DEBUG[2779] app_queue.c: Device 'SIP/0004f215aabb-1' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Jul 6 15:52:36] VERBOSE[2782] chan_sip.c: <--- SIP read from UDP:10.0.5.198:5060 ---> REGISTER sip:10.0.5.205 SIP/2.0 Via: SIP/2.0/UDP 10.0.5.198;branch=z9hG4bK4d59cc75DF178684 From: "0004f215aabb-2" ;tag=9B3D1B49-FDDEE0E6 To: CSeq: 11 REGISTER Call-ID: 5f6fe385-15b70ccf-d62de2d4@10.0.5.198 Contact: ;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER" User-Agent: PolycomSoundPointIP-SPIP_501-UA/3.1.6.0017 Accept-Language: fr-fr,fr;q=0.9,en;q=0.8 Authorization: Digest username="0004f215aabb-2", realm="asterisk", nonce="509eb1a0", uri="sip:10.0.5.205", response="7327e8f113887adee494a1f66e9f34bc", algorithm=MD5 Max-Forwards: 70 Expires: 60 Content-Length: 0 <-------------> [Jul 6 15:52:36] DEBUG[2782] chan_sip.c: Header 0 [ 31]: REGISTER sip:10.0.5.205 SIP/2.0 [Jul 6 15:52:36] DEBUG[2782] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 10.0.5.198;branch=z9hG4bK4d59cc75DF178684 [Jul 6 15:52:36] DEBUG[2782] chan_sip.c: Header 2 [ 76]: From: "0004f215aabb-2" ;tag=9B3D1B49-FDDEE0E6 [Jul 6 15:52:36] DEBUG[2782] chan_sip.c: Header 3 [ 35]: To: [Jul 6 15:52:36] DEBUG[2782] chan_sip.c: Header 4 [ 17]: CSeq: 11 REGISTER [Jul 6 15:52:36] DEBUG[2782] chan_sip.c: Header 5 [ 46]: Call-ID: 5f6fe385-15b70ccf-d62de2d4@10.0.5.198 [Jul 6 15:52:36] DEBUG[2782] chan_sip.c: Header 6 [140]: Contact: ;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER" [Jul 6 15:52:36] DEBUG[2782] chan_sip.c: Header 7 [ 54]: User-Agent: PolycomSoundPointIP-SPIP_501-UA/3.1.6.0017 [Jul 6 15:52:36] DEBUG[2782] chan_sip.c: Header 8 [ 40]: Accept-Language: fr-fr,fr;q=0.9,en;q=0.8 [Jul 6 15:52:36] DEBUG[2782] chan_sip.c: Header 9 [165]: Authorization: Digest username="0004f215aabb-2", realm="asterisk", nonce="509eb1a0", uri="sip:10.0.5.205", response="7327e8f113887adee494a1f66e9f34bc", algorithm=MD5 [Jul 6 15:52:36] DEBUG[2782] chan_sip.c: Header 10 [ 16]: Max-Forwards: 70 [Jul 6 15:52:36] DEBUG[2782] chan_sip.c: Header 11 [ 11]: Expires: 60 [Jul 6 15:52:36] DEBUG[2782] chan_sip.c: Header 12 [ 17]: Content-Length: 0 [Jul 6 15:52:36] DEBUG[2782] chan_sip.c: Header 13 [ 0]: [Jul 6 15:52:36] VERBOSE[2782] chan_sip.c: --- (13 headers 0 lines) --- [Jul 6 15:52:36] DEBUG[2782] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Jul 6 15:52:36] DEBUG[2782] chan_sip.c: Initializing initreq for method REGISTER - callid 5f6fe385-15b70ccf-d62de2d4@10.0.5.198 [Jul 6 15:52:36] VERBOSE[2782] chan_sip.c: Sending to 10.0.5.198 : 5060 (no NAT) [Jul 6 15:52:36] NOTICE[2782] chan_sip.c: Correct auth, but based on stale nonce received from '' [Jul 6 15:52:36] VERBOSE[2782] chan_sip.c: <--- Transmitting (no NAT) to 10.0.5.198:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.0.5.198;branch=z9hG4bK4d59cc75DF178684;received=10.0.5.198 From: "0004f215aabb-2" ;tag=9B3D1B49-FDDEE0E6 To: ;tag=as3ebeda14 Call-ID: 5f6fe385-15b70ccf-d62de2d4@10.0.5.198 CSeq: 11 REGISTER Server: Asterisk PBX 1.6.2.9 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="791bf189", stale=true Content-Length: 0 <------------> [Jul 6 15:52:36] DEBUG[2782] chan_sip.c: Trying to put 'SIP/2.0 401' onto UDP socket destined for 10.0.5.198:5060 [Jul 6 15:52:36] VERBOSE[2782] chan_sip.c: Scheduling destruction of SIP dialog '5f6fe385-15b70ccf-d62de2d4@10.0.5.198' in 32000 ms (Method: REGISTER) [Jul 6 15:52:36] VERBOSE[2782] chan_sip.c: <--- SIP read from UDP:10.0.5.198:5060 ---> REGISTER sip:10.0.5.205 SIP/2.0 Via: SIP/2.0/UDP 10.0.5.198;branch=z9hG4bK4055a23f84546B16 From: "0004f215aabb-2" ;tag=9B3D1B49-FDDEE0E6 To: CSeq: 12 REGISTER Call-ID: 5f6fe385-15b70ccf-d62de2d4@10.0.5.198 Contact: ;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER" User-Agent: PolycomSoundPointIP-SPIP_501-UA/3.1.6.0017 Accept-Language: fr-fr,fr;q=0.9,en;q=0.8 Authorization: Digest username="0004f215aabb-2", realm="asterisk", nonce="791bf189", uri="sip:10.0.5.205", response="a150ca34dc66c72651d75da25bbef07b", algorithm=MD5 Max-Forwards: 70 Expires: 60 Content-Length: 0 <-------------> [Jul 6 15:52:36] DEBUG[2782] chan_sip.c: Header 0 [ 31]: REGISTER sip:10.0.5.205 SIP/2.0 [Jul 6 15:52:36] DEBUG[2782] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 10.0.5.198;branch=z9hG4bK4055a23f84546B16 [Jul 6 15:52:36] DEBUG[2782] chan_sip.c: Header 2 [ 76]: From: "0004f215aabb-2" ;tag=9B3D1B49-FDDEE0E6 [Jul 6 15:52:36] DEBUG[2782] chan_sip.c: Header 3 [ 35]: To: [Jul 6 15:52:36] DEBUG[2782] chan_sip.c: Header 4 [ 17]: CSeq: 12 REGISTER [Jul 6 15:52:36] DEBUG[2782] chan_sip.c: Header 5 [ 46]: Call-ID: 5f6fe385-15b70ccf-d62de2d4@10.0.5.198 [Jul 6 15:52:36] DEBUG[2782] chan_sip.c: Header 6 [140]: Contact: ;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER" [Jul 6 15:52:36] DEBUG[2782] chan_sip.c: Header 7 [ 54]: User-Agent: PolycomSoundPointIP-SPIP_501-UA/3.1.6.0017 [Jul 6 15:52:36] DEBUG[2782] chan_sip.c: Header 8 [ 40]: Accept-Language: fr-fr,fr;q=0.9,en;q=0.8 [Jul 6 15:52:36] DEBUG[2782] chan_sip.c: Header 9 [165]: Authorization: Digest username="0004f215aabb-2", realm="asterisk", nonce="791bf189", uri="sip:10.0.5.205", response="a150ca34dc66c72651d75da25bbef07b", algorithm=MD5 [Jul 6 15:52:36] DEBUG[2782] chan_sip.c: Header 10 [ 16]: Max-Forwards: 70 [Jul 6 15:52:36] DEBUG[2782] chan_sip.c: Header 11 [ 11]: Expires: 60 [Jul 6 15:52:36] DEBUG[2782] chan_sip.c: Header 12 [ 17]: Content-Length: 0 [Jul 6 15:52:36] DEBUG[2782] chan_sip.c: Header 13 [ 0]: [Jul 6 15:52:36] VERBOSE[2782] chan_sip.c: --- (13 headers 0 lines) --- [Jul 6 15:52:36] DEBUG[2782] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Jul 6 15:52:36] DEBUG[2782] chan_sip.c: Initializing initreq for method REGISTER - callid 5f6fe385-15b70ccf-d62de2d4@10.0.5.198 [Jul 6 15:52:36] VERBOSE[2782] chan_sip.c: Sending to 10.0.5.198 : 5060 (no NAT) [Jul 6 15:52:36] VERBOSE[2782] chan_sip.c: <--- Transmitting (no NAT) to 10.0.5.198:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.5.198;branch=z9hG4bK4055a23f84546B16;received=10.0.5.198 From: "0004f215aabb-2" ;tag=9B3D1B49-FDDEE0E6 To: ;tag=as3ebeda14 Call-ID: 5f6fe385-15b70ccf-d62de2d4@10.0.5.198 CSeq: 12 REGISTER Server: Asterisk PBX 1.6.2.9 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Expires: 60 Contact: ;expires=60 Date: Tue, 06 Jul 2010 19:52:36 GMT Content-Length: 0 <------------> [Jul 6 15:52:36] DEBUG[2782] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.0.5.198:5060 [Jul 6 15:52:36] VERBOSE[2782] chan_sip.c: Scheduling destruction of SIP dialog '5f6fe385-15b70ccf-d62de2d4@10.0.5.198' in 32000 ms (Method: REGISTER) [Jul 6 15:52:36] DEBUG[2689] devicestate.c: No provider found, checking channel drivers for SIP - 0004f215aabb-2 [Jul 6 15:52:36] DEBUG[2689] chan_sip.c: Checking device state for peer 0004f215aabb-2 [Jul 6 15:52:36] DEBUG[2689] devicestate.c: Changing state for SIP/0004f215aabb-2 - state 1 (Not in use) [Jul 6 15:52:36] DEBUG[2689] devicestate.c: device 'SIP/0004f215aabb-2' state '1' [Jul 6 15:52:36] DEBUG[2779] app_queue.c: Device 'SIP/0004f215aabb-2' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Jul 6 15:52:37] DEBUG[2782] chan_sip.c: Auto destroying SIP dialog 'c8ce19c6966b23d9' [Jul 6 15:52:37] DEBUG[2782] chan_sip.c: Destroying SIP dialog c8ce19c6966b23d9 [Jul 6 15:52:37] VERBOSE[2782] chan_sip.c: Really destroying SIP dialog 'c8ce19c6966b23d9' Method: REGISTER [Jul 6 15:52:39] DEBUG[2695] rtp.c: Got RTCP report of 64 bytes [Jul 6 15:52:44] DEBUG[2695] rtp.c: Got RTCP report of 64 bytes [Jul 6 15:52:49] DEBUG[2695] rtp.c: Got RTCP report of 64 bytes [Jul 6 15:52:50] VERBOSE[2782] chan_sip.c: <--- SIP read from UDP:10.0.5.196:5060 ---> REGISTER sip:10.0.5.191:5060 SIP/2.0 Via: SIP/2.0/UDP 10.0.5.196:5060;branch=z9hG4bK67e414e1ae279caa2.4cedbd68efa2bdb70 Max-Forwards: 70 From: ;tag=d191d653b3 To: Call-ID: c8ce19c6966b23d9 CSeq: 26482 REGISTER Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Authorization: Digest username="33i",realm="asterisk",nonce="251d90bf",uri="sip:10.0.5.191:5060",response="0069b554a4f240f71e4c115d67ef231e",algorithm=MD5 Contact: "33i" ;expires=60;+sip.instance="" Supported: gruu, path User-Agent: Aastra 9143i/2.5.3.2002 Content-Length: 0 <-------------> [Jul 6 15:52:50] DEBUG[2782] chan_sip.c: Header 0 [ 36]: REGISTER sip:10.0.5.191:5060 SIP/2.0 [Jul 6 15:52:50] DEBUG[2782] chan_sip.c: Header 1 [ 82]: Via: SIP/2.0/UDP 10.0.5.196:5060;branch=z9hG4bK67e414e1ae279caa2.4cedbd68efa2bdb70 [Jul 6 15:52:50] DEBUG[2782] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jul 6 15:52:50] DEBUG[2782] chan_sip.c: Header 3 [ 46]: From: ;tag=d191d653b3 [Jul 6 15:52:50] DEBUG[2782] chan_sip.c: Header 4 [ 29]: To: [Jul 6 15:52:50] DEBUG[2782] chan_sip.c: Header 5 [ 25]: Call-ID: c8ce19c6966b23d9 [Jul 6 15:52:50] DEBUG[2782] chan_sip.c: Header 6 [ 20]: CSeq: 26482 REGISTER [Jul 6 15:52:50] DEBUG[2782] chan_sip.c: Header 7 [ 88]: Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO [Jul 6 15:52:50] DEBUG[2782] chan_sip.c: Header 8 [ 53]: Allow-Events: talk, hold, conference, LocalModeStatus [Jul 6 15:52:50] DEBUG[2782] chan_sip.c: Header 9 [154]: Authorization: Digest username="33i",realm="asterisk",nonce="251d90bf",uri="sip:10.0.5.191:5060",response="0069b554a4f240f71e4c115d67ef231e",algorithm=MD5 [Jul 6 15:52:50] DEBUG[2782] chan_sip.c: Header 10 [129]: Contact: "33i" ;expires=60;+sip.instance="" [Jul 6 15:52:50] DEBUG[2782] chan_sip.c: Header 11 [ 21]: Supported: gruu, path [Jul 6 15:52:50] DEBUG[2782] chan_sip.c: Header 12 [ 35]: User-Agent: Aastra 9143i/2.5.3.2002 [Jul 6 15:52:50] DEBUG[2782] chan_sip.c: Header 13 [ 17]: Content-Length: 0 [Jul 6 15:52:50] DEBUG[2782] chan_sip.c: Header 14 [ 0]: [Jul 6 15:52:50] VERBOSE[2782] chan_sip.c: --- (14 headers 0 lines) --- [Jul 6 15:52:50] DEBUG[2782] acl.c: Found IP address for this socket [Jul 6 15:52:50] DEBUG[2782] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 10.0.5.191:5060 [Jul 6 15:52:50] DEBUG[2782] chan_sip.c: Allocating new SIP dialog for c8ce19c6966b23d9 - REGISTER (No RTP) [Jul 6 15:52:50] DEBUG[2782] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Jul 6 15:52:50] DEBUG[2782] chan_sip.c: Initializing initreq for method REGISTER - callid c8ce19c6966b23d9 [Jul 6 15:52:50] VERBOSE[2782] chan_sip.c: Sending to 10.0.5.196 : 5060 (no NAT) [Jul 6 15:52:50] VERBOSE[2782] chan_sip.c: <--- Transmitting (no NAT) to 10.0.5.196:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.0.5.196:5060;branch=z9hG4bK67e414e1ae279caa2.4cedbd68efa2bdb70;received=10.0.5.196 From: ;tag=d191d653b3 To: ;tag=as1b940506 Call-ID: c8ce19c6966b23d9 CSeq: 26482 REGISTER Server: Asterisk PBX 1.6.2.9 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5eb88395" Content-Length: 0 <------------> [Jul 6 15:52:50] DEBUG[2782] chan_sip.c: Trying to put 'SIP/2.0 401' onto UDP socket destined for 10.0.5.196:5060 [Jul 6 15:52:50] VERBOSE[2782] chan_sip.c: Scheduling destruction of SIP dialog 'c8ce19c6966b23d9' in 32000 ms (Method: REGISTER) [Jul 6 15:52:50] VERBOSE[2782] chan_sip.c: <--- SIP read from UDP:10.0.5.196:5060 ---> REGISTER sip:10.0.5.191:5060 SIP/2.0 Via: SIP/2.0/UDP 10.0.5.196:5060;branch=z9hG4bK9861d1899d42d24a7.4837eb2f819eda9fe Max-Forwards: 70 From: ;tag=d191d653b3 To: Call-ID: c8ce19c6966b23d9 CSeq: 26483 REGISTER Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Authorization: Digest username="33i",realm="asterisk",nonce="5eb88395",uri="sip:10.0.5.191:5060",response="96c31d281aac59efb663ce41b67308b6",algorithm=MD5 Contact: "33i" ;expires=60;+sip.instance="" Supported: gruu, path User-Agent: Aastra 9143i/2.5.3.2002 Content-Length: 0 <-------------> [Jul 6 15:52:50] DEBUG[2782] chan_sip.c: Header 0 [ 36]: REGISTER sip:10.0.5.191:5060 SIP/2.0 [Jul 6 15:52:50] DEBUG[2782] chan_sip.c: Header 1 [ 82]: Via: SIP/2.0/UDP 10.0.5.196:5060;branch=z9hG4bK9861d1899d42d24a7.4837eb2f819eda9fe [Jul 6 15:52:50] DEBUG[2782] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jul 6 15:52:50] DEBUG[2782] chan_sip.c: Header 3 [ 46]: From: ;tag=d191d653b3 [Jul 6 15:52:50] DEBUG[2782] chan_sip.c: Header 4 [ 29]: To: [Jul 6 15:52:50] DEBUG[2782] chan_sip.c: Header 5 [ 25]: Call-ID: c8ce19c6966b23d9 [Jul 6 15:52:50] DEBUG[2782] chan_sip.c: Header 6 [ 20]: CSeq: 26483 REGISTER [Jul 6 15:52:50] DEBUG[2782] chan_sip.c: Header 7 [ 88]: Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO [Jul 6 15:52:50] DEBUG[2782] chan_sip.c: Header 8 [ 53]: Allow-Events: talk, hold, conference, LocalModeStatus [Jul 6 15:52:50] DEBUG[2782] chan_sip.c: Header 9 [154]: Authorization: Digest username="33i",realm="asterisk",nonce="5eb88395",uri="sip:10.0.5.191:5060",response="96c31d281aac59efb663ce41b67308b6",algorithm=MD5 [Jul 6 15:52:50] DEBUG[2782] chan_sip.c: Header 10 [129]: Contact: "33i" ;expires=60;+sip.instance="" [Jul 6 15:52:50] DEBUG[2782] chan_sip.c: Header 11 [ 21]: Supported: gruu, path [Jul 6 15:52:50] DEBUG[2782] chan_sip.c: Header 12 [ 35]: User-Agent: Aastra 9143i/2.5.3.2002 [Jul 6 15:52:50] DEBUG[2782] chan_sip.c: Header 13 [ 17]: Content-Length: 0 [Jul 6 15:52:50] DEBUG[2782] chan_sip.c: Header 14 [ 0]: [Jul 6 15:52:50] VERBOSE[2782] chan_sip.c: --- (14 headers 0 lines) --- [Jul 6 15:52:50] DEBUG[2782] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Jul 6 15:52:50] DEBUG[2782] chan_sip.c: Initializing initreq for method REGISTER - callid c8ce19c6966b23d9 [Jul 6 15:52:50] VERBOSE[2782] chan_sip.c: Sending to 10.0.5.196 : 5060 (no NAT) [Jul 6 15:52:50] VERBOSE[2782] chan_sip.c: <--- Transmitting (no NAT) to 10.0.5.196:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.5.196:5060;branch=z9hG4bK9861d1899d42d24a7.4837eb2f819eda9fe;received=10.0.5.196 From: ;tag=d191d653b3 To: ;tag=as1b940506 Call-ID: c8ce19c6966b23d9 CSeq: 26483 REGISTER Server: Asterisk PBX 1.6.2.9 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Expires: 60 Contact: ;expires=60 Date: Tue, 06 Jul 2010 19:52:50 GMT Content-Length: 0 <------------> [Jul 6 15:52:50] DEBUG[2782] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.0.5.196:5060 [Jul 6 15:52:50] VERBOSE[2782] chan_sip.c: Scheduling destruction of SIP dialog 'c8ce19c6966b23d9' in 32000 ms (Method: REGISTER) [Jul 6 15:52:50] DEBUG[2689] devicestate.c: No provider found, checking channel drivers for SIP - 33i [Jul 6 15:52:50] DEBUG[2689] chan_sip.c: Checking device state for peer 33i [Jul 6 15:52:50] DEBUG[2689] devicestate.c: Changing state for SIP/33i - state 1 (Not in use) [Jul 6 15:52:50] DEBUG[2689] devicestate.c: device 'SIP/33i' state '1' [Jul 6 15:52:50] DEBUG[2779] app_queue.c: Device 'SIP/33i' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Jul 6 15:52:54] DEBUG[2695] rtp.c: Got RTCP report of 64 bytes [Jul 6 15:52:57] DEBUG[2782] chan_sip.c: Auto destroying SIP dialog '4bdb216169a15be718a35d1f6247c7b8@10.0.5.191' [Jul 6 15:52:57] DEBUG[2782] chan_sip.c: Destroying SIP dialog 4bdb216169a15be718a35d1f6247c7b8@10.0.5.191 [Jul 6 15:52:57] VERBOSE[2782] chan_sip.c: Really destroying SIP dialog '4bdb216169a15be718a35d1f6247c7b8@10.0.5.191' Method: BYE [Jul 6 15:52:57] DEBUG[2782] chan_sip.c: Updating call counter for outgoing call [Jul 6 15:52:57] DEBUG[2782] chan_sip.c: Call to peer '0004f215aabb-1' removed from call limit 0 [Jul 6 15:52:57] DEBUG[2782] chan_sip.c: This call did not properly clean up call limits. Call ID 4bdb216169a15be718a35d1f6247c7b8@10.0.5.191 [Jul 6 15:52:57] DEBUG[2689] devicestate.c: No provider found, checking channel drivers for SIP - 0004f215aabb-1 [Jul 6 15:52:57] DEBUG[2689] chan_sip.c: Checking device state for peer 0004f215aabb-1 [Jul 6 15:52:57] DEBUG[2689] devicestate.c: Changing state for SIP/0004f215aabb-1 - state 1 (Not in use) [Jul 6 15:52:57] DEBUG[2689] devicestate.c: device 'SIP/0004f215aabb-1' state '1' [Jul 6 15:52:57] DEBUG[2779] app_queue.c: Device 'SIP/0004f215aabb-1' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Jul 6 15:52:57] VERBOSE[2782] chan_sip.c: <--- SIP read from UDP:10.0.5.198:5060 ---> SUBSCRIBE sip:705@10.0.5.191 SIP/2.0 Via: SIP/2.0/UDP 10.0.5.198;branch=z9hG4bKe6b259395A893098 From: "0004f215aabb-1" ;tag=A4C8AA47-8D60C6C To: CSeq: 1 SUBSCRIBE Call-ID: 16220be3-a307e0bd-5d14c38a@10.0.5.198 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER Event: presence User-Agent: PolycomSoundPointIP-SPIP_501-UA/3.1.6.0017 Accept-Language: fr-fr,fr;q=0.9,en;q=0.8 Accept: application/xpidf+xml,text/xml+msrtc.pidf Max-Forwards: 70 Expires: 3600 Content-Length: 0 <-------------> [Jul 6 15:52:57] DEBUG[2782] chan_sip.c: Header 0 [ 36]: SUBSCRIBE sip:705@10.0.5.191 SIP/2.0 [Jul 6 15:52:57] DEBUG[2782] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 10.0.5.198;branch=z9hG4bKe6b259395A893098 [Jul 6 15:52:57] DEBUG[2782] chan_sip.c: Header 2 [ 75]: From: "0004f215aabb-1" ;tag=A4C8AA47-8D60C6C [Jul 6 15:52:57] DEBUG[2782] chan_sip.c: Header 3 [ 24]: To: [Jul 6 15:52:57] DEBUG[2782] chan_sip.c: Header 4 [ 17]: CSeq: 1 SUBSCRIBE [Jul 6 15:52:57] DEBUG[2782] chan_sip.c: Header 5 [ 46]: Call-ID: 16220be3-a307e0bd-5d14c38a@10.0.5.198 [Jul 6 15:52:57] DEBUG[2782] chan_sip.c: Header 6 [ 40]: Contact: [Jul 6 15:52:57] DEBUG[2782] chan_sip.c: Header 7 [ 96]: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER [Jul 6 15:52:57] DEBUG[2782] chan_sip.c: Header 8 [ 15]: Event: presence [Jul 6 15:52:57] DEBUG[2782] chan_sip.c: Header 9 [ 54]: User-Agent: PolycomSoundPointIP-SPIP_501-UA/3.1.6.0017 [Jul 6 15:52:57] DEBUG[2782] chan_sip.c: Header 10 [ 40]: Accept-Language: fr-fr,fr;q=0.9,en;q=0.8 [Jul 6 15:52:57] DEBUG[2782] chan_sip.c: Header 11 [ 49]: Accept: application/xpidf+xml,text/xml+msrtc.pidf [Jul 6 15:52:57] DEBUG[2782] chan_sip.c: Header 12 [ 16]: Max-Forwards: 70 [Jul 6 15:52:57] DEBUG[2782] chan_sip.c: Header 13 [ 13]: Expires: 3600 [Jul 6 15:52:57] DEBUG[2782] chan_sip.c: Header 14 [ 17]: Content-Length: 0 [Jul 6 15:52:57] DEBUG[2782] chan_sip.c: Header 15 [ 0]: [Jul 6 15:52:57] VERBOSE[2782] chan_sip.c: --- (15 headers 0 lines) --- [Jul 6 15:52:57] DEBUG[2782] acl.c: Found IP address for this socket [Jul 6 15:52:57] DEBUG[2782] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 10.0.5.191:5060 [Jul 6 15:52:57] DEBUG[2782] chan_sip.c: Allocating new SIP dialog for 16220be3-a307e0bd-5d14c38a@10.0.5.198 - SUBSCRIBE (No RTP) [Jul 6 15:52:57] DEBUG[2782] chan_sip.c: **** Received SUBSCRIBE (10) - Command in SIP SUBSCRIBE [Jul 6 15:52:57] VERBOSE[2782] chan_sip.c: Creating new subscription [Jul 6 15:52:57] DEBUG[2782] chan_sip.c: Initializing initreq for method SUBSCRIBE - callid 16220be3-a307e0bd-5d14c38a@10.0.5.198 [Jul 6 15:52:57] VERBOSE[2782] chan_sip.c: Sending to 10.0.5.198 : 5060 (no NAT) [Jul 6 15:52:57] DEBUG[2782] chan_sip.c: build_route: Contact hop: [Jul 6 15:52:57] VERBOSE[2782] chan_sip.c: list_route: hop: [Jul 6 15:52:57] VERBOSE[2782] chan_sip.c: Found peer '0004f215aabb-1' for '0004f215aabb-1' from 10.0.5.198:5060 [Jul 6 15:52:57] VERBOSE[2782] chan_sip.c: <--- Transmitting (no NAT) to 10.0.5.198:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.0.5.198;branch=z9hG4bKe6b259395A893098;received=10.0.5.198 From: "0004f215aabb-1" ;tag=A4C8AA47-8D60C6C To: ;tag=as62cfc6b2 Call-ID: 16220be3-a307e0bd-5d14c38a@10.0.5.198 CSeq: 1 SUBSCRIBE Server: Asterisk PBX 1.6.2.9 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4179540e" Content-Length: 0 <------------> [Jul 6 15:52:57] DEBUG[2782] chan_sip.c: Trying to put 'SIP/2.0 401' onto UDP socket destined for 10.0.5.198:5060 [Jul 6 15:52:57] VERBOSE[2782] chan_sip.c: Scheduling destruction of SIP dialog '16220be3-a307e0bd-5d14c38a@10.0.5.198' in 32000 ms (Method: SUBSCRIBE) [Jul 6 15:52:57] VERBOSE[2782] chan_sip.c: <--- SIP read from UDP:10.0.5.198:5060 ---> SUBSCRIBE sip:705@10.0.5.191 SIP/2.0 Via: SIP/2.0/UDP 10.0.5.198;branch=z9hG4bK42790fbeB78E8701 From: "0004f215aabb-1" ;tag=A4C8AA47-8D60C6C To: CSeq: 2 SUBSCRIBE Call-ID: 16220be3-a307e0bd-5d14c38a@10.0.5.198 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER Event: presence User-Agent: PolycomSoundPointIP-SPIP_501-UA/3.1.6.0017 Accept-Language: fr-fr,fr;q=0.9,en;q=0.8 Accept: application/xpidf+xml,text/xml+msrtc.pidf Authorization: Digest username="0004f215aabb-1", realm="asterisk", nonce="4179540e", uri="sip:705@10.0.5.191", response="e8b6ed50e23e7b7f0d6bad37f8189fb2", algorithm=MD5 Max-Forwards: 70 Expires: 3600 Content-Length: 0 <-------------> [Jul 6 15:52:57] DEBUG[2782] chan_sip.c: Header 0 [ 36]: SUBSCRIBE sip:705@10.0.5.191 SIP/2.0 [Jul 6 15:52:57] DEBUG[2782] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 10.0.5.198;branch=z9hG4bK42790fbeB78E8701 [Jul 6 15:52:57] DEBUG[2782] chan_sip.c: Header 2 [ 75]: From: "0004f215aabb-1" ;tag=A4C8AA47-8D60C6C [Jul 6 15:52:57] DEBUG[2782] chan_sip.c: Header 3 [ 24]: To: [Jul 6 15:52:57] DEBUG[2782] chan_sip.c: Header 4 [ 17]: CSeq: 2 SUBSCRIBE [Jul 6 15:52:57] DEBUG[2782] chan_sip.c: Header 5 [ 46]: Call-ID: 16220be3-a307e0bd-5d14c38a@10.0.5.198 [Jul 6 15:52:57] DEBUG[2782] chan_sip.c: Header 6 [ 40]: Contact: [Jul 6 15:52:57] DEBUG[2782] chan_sip.c: Header 7 [ 96]: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER [Jul 6 15:52:57] DEBUG[2782] chan_sip.c: Header 8 [ 15]: Event: presence [Jul 6 15:52:57] DEBUG[2782] chan_sip.c: Header 9 [ 54]: User-Agent: PolycomSoundPointIP-SPIP_501-UA/3.1.6.0017 [Jul 6 15:52:57] DEBUG[2782] chan_sip.c: Header 10 [ 40]: Accept-Language: fr-fr,fr;q=0.9,en;q=0.8 [Jul 6 15:52:57] DEBUG[2782] chan_sip.c: Header 11 [ 49]: Accept: application/xpidf+xml,text/xml+msrtc.pidf [Jul 6 15:52:57] DEBUG[2782] chan_sip.c: Header 12 [169]: Authorization: Digest username="0004f215aabb-1", realm="asterisk", nonce="4179540e", uri="sip:705@10.0.5.191", response="e8b6ed50e23e7b7f0d6bad37f8189fb2", algorithm=MD5 [Jul 6 15:52:57] DEBUG[2782] chan_sip.c: Header 13 [ 16]: Max-Forwards: 70 [Jul 6 15:52:57] DEBUG[2782] chan_sip.c: Header 14 [ 13]: Expires: 3600 [Jul 6 15:52:57] DEBUG[2782] chan_sip.c: Header 15 [ 17]: Content-Length: 0 [Jul 6 15:52:57] DEBUG[2782] chan_sip.c: Header 16 [ 0]: [Jul 6 15:52:57] VERBOSE[2782] chan_sip.c: --- (16 headers 0 lines) --- [Jul 6 15:52:57] DEBUG[2782] chan_sip.c: **** Received SUBSCRIBE (10) - Command in SIP SUBSCRIBE [Jul 6 15:52:57] DEBUG[2782] chan_sip.c: Got a new subscription 16220be3-a307e0bd-5d14c38a@10.0.5.198 (possibly with auth) [Jul 6 15:52:57] VERBOSE[2782] chan_sip.c: Creating new subscription [Jul 6 15:52:57] DEBUG[2782] chan_sip.c: Initializing initreq for method SUBSCRIBE - callid 16220be3-a307e0bd-5d14c38a@10.0.5.198 [Jul 6 15:52:57] VERBOSE[2782] chan_sip.c: Sending to 10.0.5.198 : 5060 (no NAT) [Jul 6 15:52:57] DEBUG[2782] chan_sip.c: build_route: Retaining previous route: [Jul 6 15:52:57] VERBOSE[2782] chan_sip.c: Found peer '0004f215aabb-1' for '0004f215aabb-1' from 10.0.5.198:5060 [Jul 6 15:52:57] VERBOSE[2782] chan_sip.c: Looking for 705 in 5555555555-hints (domain 10.0.5.191) [Jul 6 15:52:57] VERBOSE[2782] chan_sip.c: <--- Transmitting (no NAT) to 10.0.5.198:5060 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP 10.0.5.198;branch=z9hG4bK42790fbeB78E8701;received=10.0.5.198 From: "0004f215aabb-1" ;tag=A4C8AA47-8D60C6C To: ;tag=as62cfc6b2 Call-ID: 16220be3-a307e0bd-5d14c38a@10.0.5.198 CSeq: 2 SUBSCRIBE Server: Asterisk PBX 1.6.2.9 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 <------------> [Jul 6 15:52:57] DEBUG[2782] chan_sip.c: Trying to put 'SIP/2.0 404' onto UDP socket destined for 10.0.5.198:5060 [Jul 6 15:52:57] DEBUG[2782] chan_sip.c: Destroying SIP dialog 16220be3-a307e0bd-5d14c38a@10.0.5.198 [Jul 6 15:52:57] VERBOSE[2782] chan_sip.c: Really destroying SIP dialog '16220be3-a307e0bd-5d14c38a@10.0.5.198' Method: SUBSCRIBE [Jul 6 15:52:59] DEBUG[2695] rtp.c: Got RTCP report of 64 bytes [Jul 6 15:53:04] DEBUG[2695] rtp.c: Got RTCP report of 64 bytes [Jul 6 15:53:05] VERBOSE[2782] chan_sip.c: <--- SIP read from UDP:10.0.5.198:5060 ---> REGISTER sip:10.0.5.191 SIP/2.0 Via: SIP/2.0/UDP 10.0.5.198;branch=z9hG4bKea27816b7535A3B2 From: "0004f215aabb-1" ;tag=6F5D75EC-D185443D To: CSeq: 13 REGISTER Call-ID: ad2a1218-82f9b90a-393cb63@10.0.5.198 Contact: ;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER" User-Agent: PolycomSoundPointIP-SPIP_501-UA/3.1.6.0017 Accept-Language: fr-fr,fr;q=0.9,en;q=0.8 Authorization: Digest username="0004f215aabb-1", realm="asterisk", nonce="1961ef9a", uri="sip:10.0.5.191", response="013c587edf5c4be5b37a420fb3d73ba4", algorithm=MD5 Max-Forwards: 70 Expires: 60 Content-Length: 0 <-------------> [Jul 6 15:53:05] DEBUG[2782] chan_sip.c: Header 0 [ 31]: REGISTER sip:10.0.5.191 SIP/2.0 [Jul 6 15:53:05] DEBUG[2782] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 10.0.5.198;branch=z9hG4bKea27816b7535A3B2 [Jul 6 15:53:05] DEBUG[2782] chan_sip.c: Header 2 [ 76]: From: "0004f215aabb-1" ;tag=6F5D75EC-D185443D [Jul 6 15:53:05] DEBUG[2782] chan_sip.c: Header 3 [ 35]: To: [Jul 6 15:53:05] DEBUG[2782] chan_sip.c: Header 4 [ 17]: CSeq: 13 REGISTER [Jul 6 15:53:05] DEBUG[2782] chan_sip.c: Header 5 [ 45]: Call-ID: ad2a1218-82f9b90a-393cb63@10.0.5.198 [Jul 6 15:53:05] DEBUG[2782] chan_sip.c: Header 6 [140]: Contact: ;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER" [Jul 6 15:53:05] DEBUG[2782] chan_sip.c: Header 7 [ 54]: User-Agent: PolycomSoundPointIP-SPIP_501-UA/3.1.6.0017 [Jul 6 15:53:05] DEBUG[2782] chan_sip.c: Header 8 [ 40]: Accept-Language: fr-fr,fr;q=0.9,en;q=0.8 [Jul 6 15:53:05] DEBUG[2782] chan_sip.c: Header 9 [165]: Authorization: Digest username="0004f215aabb-1", realm="asterisk", nonce="1961ef9a", uri="sip:10.0.5.191", response="013c587edf5c4be5b37a420fb3d73ba4", algorithm=MD5 [Jul 6 15:53:05] DEBUG[2782] chan_sip.c: Header 10 [ 16]: Max-Forwards: 70 [Jul 6 15:53:05] DEBUG[2782] chan_sip.c: Header 11 [ 11]: Expires: 60 [Jul 6 15:53:05] DEBUG[2782] chan_sip.c: Header 12 [ 17]: Content-Length: 0 [Jul 6 15:53:05] DEBUG[2782] chan_sip.c: Header 13 [ 0]: [Jul 6 15:53:05] VERBOSE[2782] chan_sip.c: --- (13 headers 0 lines) --- [Jul 6 15:53:05] DEBUG[2782] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Jul 6 15:53:05] DEBUG[2782] chan_sip.c: Initializing initreq for method REGISTER - callid ad2a1218-82f9b90a-393cb63@10.0.5.198 [Jul 6 15:53:05] VERBOSE[2782] chan_sip.c: Sending to 10.0.5.198 : 5060 (no NAT) [Jul 6 15:53:05] NOTICE[2782] chan_sip.c: Correct auth, but based on stale nonce received from '' [Jul 6 15:53:05] VERBOSE[2782] chan_sip.c: <--- Transmitting (no NAT) to 10.0.5.198:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.0.5.198;branch=z9hG4bKea27816b7535A3B2;received=10.0.5.198 From: "0004f215aabb-1" ;tag=6F5D75EC-D185443D To: ;tag=as543567ce Call-ID: ad2a1218-82f9b90a-393cb63@10.0.5.198 CSeq: 13 REGISTER Server: Asterisk PBX 1.6.2.9 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6d673744", stale=true Content-Length: 0 <------------> [Jul 6 15:53:05] DEBUG[2782] chan_sip.c: Trying to put 'SIP/2.0 401' onto UDP socket destined for 10.0.5.198:5060 [Jul 6 15:53:05] VERBOSE[2782] chan_sip.c: Scheduling destruction of SIP dialog 'ad2a1218-82f9b90a-393cb63@10.0.5.198' in 32000 ms (Method: REGISTER) [Jul 6 15:53:05] VERBOSE[2782] chan_sip.c: <--- SIP read from UDP:10.0.5.198:5060 ---> REGISTER sip:10.0.5.191 SIP/2.0 Via: SIP/2.0/UDP 10.0.5.198;branch=z9hG4bK6f673005BAE0E954 From: "0004f215aabb-1" ;tag=6F5D75EC-D185443D To: CSeq: 14 REGISTER Call-ID: ad2a1218-82f9b90a-393cb63@10.0.5.198 Contact: ;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER" User-Agent: PolycomSoundPointIP-SPIP_501-UA/3.1.6.0017 Accept-Language: fr-fr,fr;q=0.9,en;q=0.8 Authorization: Digest username="0004f215aabb-1", realm="asterisk", nonce="6d673744", uri="sip:10.0.5.191", response="8f7a7b528c6893cae38d8f7b1ba3c42e", algorithm=MD5 Max-Forwards: 70 Expires: 60 Content-Length: 0 <-------------> [Jul 6 15:53:05] DEBUG[2782] chan_sip.c: Header 0 [ 31]: REGISTER sip:10.0.5.191 SIP/2.0 [Jul 6 15:53:05] DEBUG[2782] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 10.0.5.198;branch=z9hG4bK6f673005BAE0E954 [Jul 6 15:53:05] DEBUG[2782] chan_sip.c: Header 2 [ 76]: From: "0004f215aabb-1" ;tag=6F5D75EC-D185443D [Jul 6 15:53:05] DEBUG[2782] chan_sip.c: Header 3 [ 35]: To: [Jul 6 15:53:05] DEBUG[2782] chan_sip.c: Header 4 [ 17]: CSeq: 14 REGISTER [Jul 6 15:53:05] DEBUG[2782] chan_sip.c: Header 5 [ 45]: Call-ID: ad2a1218-82f9b90a-393cb63@10.0.5.198 [Jul 6 15:53:05] DEBUG[2782] chan_sip.c: Header 6 [140]: Contact: ;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER" [Jul 6 15:53:05] DEBUG[2782] chan_sip.c: Header 7 [ 54]: User-Agent: PolycomSoundPointIP-SPIP_501-UA/3.1.6.0017 [Jul 6 15:53:05] DEBUG[2782] chan_sip.c: Header 8 [ 40]: Accept-Language: fr-fr,fr;q=0.9,en;q=0.8 [Jul 6 15:53:05] DEBUG[2782] chan_sip.c: Header 9 [165]: Authorization: Digest username="0004f215aabb-1", realm="asterisk", nonce="6d673744", uri="sip:10.0.5.191", response="8f7a7b528c6893cae38d8f7b1ba3c42e", algorithm=MD5 [Jul 6 15:53:05] DEBUG[2782] chan_sip.c: Header 10 [ 16]: Max-Forwards: 70 [Jul 6 15:53:05] DEBUG[2782] chan_sip.c: Header 11 [ 11]: Expires: 60 [Jul 6 15:53:05] DEBUG[2782] chan_sip.c: Header 12 [ 17]: Content-Length: 0 [Jul 6 15:53:05] DEBUG[2782] chan_sip.c: Header 13 [ 0]: [Jul 6 15:53:05] VERBOSE[2782] chan_sip.c: --- (13 headers 0 lines) --- [Jul 6 15:53:05] DEBUG[2782] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Jul 6 15:53:05] DEBUG[2782] chan_sip.c: Initializing initreq for method REGISTER - callid ad2a1218-82f9b90a-393cb63@10.0.5.198 [Jul 6 15:53:05] VERBOSE[2782] chan_sip.c: Sending to 10.0.5.198 : 5060 (no NAT) [Jul 6 15:53:05] VERBOSE[2782] chan_sip.c: <--- Transmitting (no NAT) to 10.0.5.198:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.5.198;branch=z9hG4bK6f673005BAE0E954;received=10.0.5.198 From: "0004f215aabb-1" ;tag=6F5D75EC-D185443D To: ;tag=as543567ce Call-ID: ad2a1218-82f9b90a-393cb63@10.0.5.198 CSeq: 14 REGISTER Server: Asterisk PBX 1.6.2.9 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Expires: 60 Contact: ;expires=60 Date: Tue, 06 Jul 2010 19:53:05 GMT Content-Length: 0 <------------> [Jul 6 15:53:05] DEBUG[2782] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.0.5.198:5060 [Jul 6 15:53:05] VERBOSE[2782] chan_sip.c: Scheduling destruction of SIP dialog 'ad2a1218-82f9b90a-393cb63@10.0.5.198' in 32000 ms (Method: REGISTER) [Jul 6 15:53:05] DEBUG[2689] devicestate.c: No provider found, checking channel drivers for SIP - 0004f215aabb-1 [Jul 6 15:53:05] DEBUG[2689] chan_sip.c: Checking device state for peer 0004f215aabb-1 [Jul 6 15:53:05] DEBUG[2689] devicestate.c: Changing state for SIP/0004f215aabb-1 - state 1 (Not in use) [Jul 6 15:53:05] DEBUG[2689] devicestate.c: device 'SIP/0004f215aabb-1' state '1' [Jul 6 15:53:05] DEBUG[2779] app_queue.c: Device 'SIP/0004f215aabb-1' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Jul 6 15:53:06] VERBOSE[2782] chan_sip.c: <--- SIP read from UDP:10.0.5.198:5060 ---> REGISTER sip:10.0.5.205 SIP/2.0 Via: SIP/2.0/UDP 10.0.5.198;branch=z9hG4bKdd33554f317C9366 From: "0004f215aabb-2" ;tag=9B3D1B49-FDDEE0E6 To: CSeq: 13 REGISTER Call-ID: 5f6fe385-15b70ccf-d62de2d4@10.0.5.198 Contact: ;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER" User-Agent: PolycomSoundPointIP-SPIP_501-UA/3.1.6.0017 Accept-Language: fr-fr,fr;q=0.9,en;q=0.8 Authorization: Digest username="0004f215aabb-2", realm="asterisk", nonce="791bf189", uri="sip:10.0.5.205", response="a150ca34dc66c72651d75da25bbef07b", algorithm=MD5 Max-Forwards: 70 Expires: 60 Content-Length: 0 <-------------> [Jul 6 15:53:06] DEBUG[2782] chan_sip.c: Header 0 [ 31]: REGISTER sip:10.0.5.205 SIP/2.0 [Jul 6 15:53:06] DEBUG[2782] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 10.0.5.198;branch=z9hG4bKdd33554f317C9366 [Jul 6 15:53:06] DEBUG[2782] chan_sip.c: Header 2 [ 76]: From: "0004f215aabb-2" ;tag=9B3D1B49-FDDEE0E6 [Jul 6 15:53:06] DEBUG[2782] chan_sip.c: Header 3 [ 35]: To: [Jul 6 15:53:06] DEBUG[2782] chan_sip.c: Header 4 [ 17]: CSeq: 13 REGISTER [Jul 6 15:53:06] DEBUG[2782] chan_sip.c: Header 5 [ 46]: Call-ID: 5f6fe385-15b70ccf-d62de2d4@10.0.5.198 [Jul 6 15:53:06] DEBUG[2782] chan_sip.c: Header 6 [140]: Contact: ;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER" [Jul 6 15:53:06] DEBUG[2782] chan_sip.c: Header 7 [ 54]: User-Agent: PolycomSoundPointIP-SPIP_501-UA/3.1.6.0017 [Jul 6 15:53:06] DEBUG[2782] chan_sip.c: Header 8 [ 40]: Accept-Language: fr-fr,fr;q=0.9,en;q=0.8 [Jul 6 15:53:06] DEBUG[2782] chan_sip.c: Header 9 [165]: Authorization: Digest username="0004f215aabb-2", realm="asterisk", nonce="791bf189", uri="sip:10.0.5.205", response="a150ca34dc66c72651d75da25bbef07b", algorithm=MD5 [Jul 6 15:53:06] DEBUG[2782] chan_sip.c: Header 10 [ 16]: Max-Forwards: 70 [Jul 6 15:53:06] DEBUG[2782] chan_sip.c: Header 11 [ 11]: Expires: 60 [Jul 6 15:53:06] DEBUG[2782] chan_sip.c: Header 12 [ 17]: Content-Length: 0 [Jul 6 15:53:06] DEBUG[2782] chan_sip.c: Header 13 [ 0]: [Jul 6 15:53:06] VERBOSE[2782] chan_sip.c: --- (13 headers 0 lines) --- [Jul 6 15:53:06] DEBUG[2782] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Jul 6 15:53:06] DEBUG[2782] chan_sip.c: Initializing initreq for method REGISTER - callid 5f6fe385-15b70ccf-d62de2d4@10.0.5.198 [Jul 6 15:53:06] VERBOSE[2782] chan_sip.c: Sending to 10.0.5.198 : 5060 (no NAT) [Jul 6 15:53:06] NOTICE[2782] chan_sip.c: Correct auth, but based on stale nonce received from '' [Jul 6 15:53:06] VERBOSE[2782] chan_sip.c: <--- Transmitting (no NAT) to 10.0.5.198:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.0.5.198;branch=z9hG4bKdd33554f317C9366;received=10.0.5.198 From: "0004f215aabb-2" ;tag=9B3D1B49-FDDEE0E6 To: ;tag=as3ebeda14 Call-ID: 5f6fe385-15b70ccf-d62de2d4@10.0.5.198 CSeq: 13 REGISTER Server: Asterisk PBX 1.6.2.9 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6d06e39f", stale=true Content-Length: 0 <------------> [Jul 6 15:53:06] DEBUG[2782] chan_sip.c: Trying to put 'SIP/2.0 401' onto UDP socket destined for 10.0.5.198:5060 [Jul 6 15:53:06] VERBOSE[2782] chan_sip.c: Scheduling destruction of SIP dialog '5f6fe385-15b70ccf-d62de2d4@10.0.5.198' in 32000 ms (Method: REGISTER) [Jul 6 15:53:06] VERBOSE[2782] chan_sip.c: <--- SIP read from UDP:10.0.5.198:5060 ---> REGISTER sip:10.0.5.205 SIP/2.0 Via: SIP/2.0/UDP 10.0.5.198;branch=z9hG4bKf0237fc9AD001268 From: "0004f215aabb-2" ;tag=9B3D1B49-FDDEE0E6 To: CSeq: 14 REGISTER Call-ID: 5f6fe385-15b70ccf-d62de2d4@10.0.5.198 Contact: ;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER" User-Agent: PolycomSoundPointIP-SPIP_501-UA/3.1.6.0017 Accept-Language: fr-fr,fr;q=0.9,en;q=0.8 Authorization: Digest username="0004f215aabb-2", realm="asterisk", nonce="6d06e39f", uri="sip:10.0.5.205", response="ed700653a29cb9f16f673cb281bfdc9a", algorithm=MD5 Max-Forwards: 70 Expires: 60 Content-Length: 0 <-------------> [Jul 6 15:53:06] DEBUG[2782] chan_sip.c: Header 0 [ 31]: REGISTER sip:10.0.5.205 SIP/2.0 [Jul 6 15:53:06] DEBUG[2782] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 10.0.5.198;branch=z9hG4bKf0237fc9AD001268 [Jul 6 15:53:06] DEBUG[2782] chan_sip.c: Header 2 [ 76]: From: "0004f215aabb-2" ;tag=9B3D1B49-FDDEE0E6 [Jul 6 15:53:06] DEBUG[2782] chan_sip.c: Header 3 [ 35]: To: [Jul 6 15:53:06] DEBUG[2782] chan_sip.c: Header 4 [ 17]: CSeq: 14 REGISTER [Jul 6 15:53:06] DEBUG[2782] chan_sip.c: Header 5 [ 46]: Call-ID: 5f6fe385-15b70ccf-d62de2d4@10.0.5.198 [Jul 6 15:53:06] DEBUG[2782] chan_sip.c: Header 6 [140]: Contact: ;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER" [Jul 6 15:53:06] DEBUG[2782] chan_sip.c: Header 7 [ 54]: User-Agent: PolycomSoundPointIP-SPIP_501-UA/3.1.6.0017 [Jul 6 15:53:06] DEBUG[2782] chan_sip.c: Header 8 [ 40]: Accept-Language: fr-fr,fr;q=0.9,en;q=0.8 [Jul 6 15:53:06] DEBUG[2782] chan_sip.c: Header 9 [165]: Authorization: Digest username="0004f215aabb-2", realm="asterisk", nonce="6d06e39f", uri="sip:10.0.5.205", response="ed700653a29cb9f16f673cb281bfdc9a", algorithm=MD5 [Jul 6 15:53:06] DEBUG[2782] chan_sip.c: Header 10 [ 16]: Max-Forwards: 70 [Jul 6 15:53:06] DEBUG[2782] chan_sip.c: Header 11 [ 11]: Expires: 60 [Jul 6 15:53:06] DEBUG[2782] chan_sip.c: Header 12 [ 17]: Content-Length: 0 [Jul 6 15:53:06] DEBUG[2782] chan_sip.c: Header 13 [ 0]: [Jul 6 15:53:06] VERBOSE[2782] chan_sip.c: --- (13 headers 0 lines) --- [Jul 6 15:53:06] DEBUG[2782] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Jul 6 15:53:06] DEBUG[2782] chan_sip.c: Initializing initreq for method REGISTER - callid 5f6fe385-15b70ccf-d62de2d4@10.0.5.198 [Jul 6 15:53:06] VERBOSE[2782] chan_sip.c: Sending to 10.0.5.198 : 5060 (no NAT) [Jul 6 15:53:06] VERBOSE[2782] chan_sip.c: <--- Transmitting (no NAT) to 10.0.5.198:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.5.198;branch=z9hG4bKf0237fc9AD001268;received=10.0.5.198 From: "0004f215aabb-2" ;tag=9B3D1B49-FDDEE0E6 To: ;tag=as3ebeda14 Call-ID: 5f6fe385-15b70ccf-d62de2d4@10.0.5.198 CSeq: 14 REGISTER Server: Asterisk PBX 1.6.2.9 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Expires: 60 Contact: ;expires=60 Date: Tue, 06 Jul 2010 19:53:06 GMT Content-Length: 0 <------------> [Jul 6 15:53:06] DEBUG[2782] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.0.5.198:5060 [Jul 6 15:53:06] VERBOSE[2782] chan_sip.c: Scheduling destruction of SIP dialog '5f6fe385-15b70ccf-d62de2d4@10.0.5.198' in 32000 ms (Method: REGISTER) [Jul 6 15:53:06] DEBUG[2689] devicestate.c: No provider found, checking channel drivers for SIP - 0004f215aabb-2 [Jul 6 15:53:06] DEBUG[2689] chan_sip.c: Checking device state for peer 0004f215aabb-2 [Jul 6 15:53:06] DEBUG[2689] devicestate.c: Changing state for SIP/0004f215aabb-2 - state 1 (Not in use) [Jul 6 15:53:06] DEBUG[2689] devicestate.c: device 'SIP/0004f215aabb-2' state '1' [Jul 6 15:53:06] DEBUG[2779] app_queue.c: Device 'SIP/0004f215aabb-2' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Jul 6 15:53:09] DEBUG[2695] rtp.c: Got RTCP report of 64 bytes [Jul 6 15:53:11] DEBUG[2695] rtp.c: Setting the marker bit due to a source update [Jul 6 15:53:11] VERBOSE[2695] res_musiconhold.c: -- Stopped music on hold on SIP/33i-00000000 [Jul 6 15:53:11] DEBUG[2695] channel.c: Set channel SIP/33i-00000000 to write format ulaw [Jul 6 15:53:11] DEBUG[2695] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Jul 6 15:53:11] DEBUG[2695] pbx.c: Registered context 'park-dial'(0x871f0c0) in table 0x87661f0 registrar: features [Jul 6 15:53:11] VERBOSE[2695] pbx.c: -- Registered extension context 'park-dial' (0x871f0c0) in table 0x87661f0; registrar: features [Jul 6 15:53:11] DEBUG[2695] pbx.c: Added extension 'SIP033i' priority 1 to park-dial (0x871f0c0) [Jul 6 15:53:11] VERBOSE[2695] pbx.c: -- Added extension 'SIP033i' priority 1 to park-dial (0x871f0c0) [Jul 6 15:53:11] VERBOSE[2695] features.c: == Timeout for SIP/33i-00000000 parked on 701 (default). Returning to park-dial,SIP033i,1 [Jul 6 15:53:11] DEBUG[2695] features.c: Notification of state change to metermaids 701@parkedcalls to state 'Not in use'[Jul 6 15:53:11] DEBUG[2695] devicestate.c: device 'park:701@parkedcalls' state '1' [Jul 6 15:53:11] DEBUG[2690] chan_sip.c: Strict routing enforced for session ab3922f0-ae400522-a64ac11b@10.0.5.198 [Jul 6 15:53:11] VERBOSE[2690] chan_sip.c: set_destination: Parsing for address/port to send to [Jul 6 15:53:11] VERBOSE[2690] chan_sip.c: set_destination: set destination to 10.0.5.198, port 5060 [Jul 6 15:53:11] VERBOSE[2690] chan_sip.c: Reliably Transmitting (no NAT) to 10.0.5.198:5060: NOTIFY sip:0004f215aabb-1@10.0.5.198 SIP/2.0 Via: SIP/2.0/UDP 10.0.5.191:5060;branch=z9hG4bK0bc2b3dc;rport Max-Forwards: 70 From: ;tag=as545536d7 To: "0004f215aabb-1" ;tag=B04C9344-7C4E7635 Contact: Call-ID: ab3922f0-ae400522-a64ac11b@10.0.5.198 CSeq: 104 NOTIFY User-Agent: Asterisk PBX 1.6.2.9 Event: presence Content-Type: application/xpidf+xml Subscription-State: active Content-Length: 348
--- [Jul 6 15:53:11] DEBUG[2690] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #89 [Jul 6 15:53:11] DEBUG[2690] chan_sip.c: Trying to put 'NOTIFY sip:' onto UDP socket destined for 10.0.5.198:5060 [Jul 6 15:53:11] VERBOSE[2690] chan_sip.c: == Extension Changed 800[5555555555-hints] new state Idle for Notify User test [Jul 6 15:53:11] DEBUG[2779] app_queue.c: Device 'park:701@parkedcalls' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Jul 6 15:53:11] DEBUG[2909] pbx.c: Launching 'Dial' [Jul 6 15:53:11] VERBOSE[2909] pbx.c: -- Executing [SIP033i@park-dial:1] Dial("SIP/33i-00000000", "SIP/33i,30,") in new stack [Jul 6 15:53:11] DEBUG[2909] chan_sip.c: Asked to create a SIP channel with formats: 0x4 (ulaw) [Jul 6 15:53:11] VERBOSE[2909] netsock.c: == Using SIP RTP CoS mark 5 [Jul 6 15:53:11] DEBUG[2909] chan_sip.c: Allocating new SIP dialog for 6b2cf410509c557d65b5cce93fad30b5@127.0.0.1 - INVITE (With RTP) [Jul 6 15:53:11] DEBUG[2909] chan_sip.c: Setting NAT on RTP to Off [Jul 6 15:53:11] DEBUG[2909] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Jul 6 15:53:11] DEBUG[2909] acl.c: Found IP address for this socket [Jul 6 15:53:11] DEBUG[2909] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 10.0.5.191:5060 [Jul 6 15:53:11] DEBUG[2909] chan_sip.c: *** Our native formats are 0x4 (ulaw) [Jul 6 15:53:11] DEBUG[2909] chan_sip.c: *** Joint capabilities are 0x4 (ulaw) [Jul 6 15:53:11] DEBUG[2909] chan_sip.c: *** Our capabilities are 0x4 (ulaw) [Jul 6 15:53:11] DEBUG[2909] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x4 (ulaw) [Jul 6 15:53:11] DEBUG[2909] chan_sip.c: *** Our preferred formats from the incoming channel are 0x4 (ulaw) [Jul 6 15:53:11] DEBUG[2909] chan_sip.c: This channel will not be able to handle video. [Jul 6 15:53:11] DEBUG[2909] rtp.c: Seeded SDP of 'SIP/33i-00000002' with that of 'SIP/33i-00000000' [Jul 6 15:53:11] DEBUG[2909] channel.c: Not copying variable DIALEDTIME. [Jul 6 15:53:11] DEBUG[2909] channel.c: Not copying variable ANSWEREDTIME. [Jul 6 15:53:11] DEBUG[2909] channel.c: Not copying variable DIALEDPEERNAME. [Jul 6 15:53:11] DEBUG[2909] channel.c: Not copying variable DIALEDPEERNUMBER. [Jul 6 15:53:11] DEBUG[2909] channel.c: Not copying variable DIALSTATUS. [Jul 6 15:53:11] DEBUG[2909] channel.c: Copying soft-transferable variable SIPTRANSFER_REFERER. [Jul 6 15:53:11] DEBUG[2909] channel.c: Copying soft-transferable variable SIPTRANSFER. [Jul 6 15:53:11] DEBUG[2909] channel.c: Not copying variable SIPDOMAIN. [Jul 6 15:53:11] DEBUG[2909] channel.c: Not copying variable BLINDTRANSFER. [Jul 6 15:53:11] DEBUG[2909] channel.c: Not copying variable SIPREFERREDBYHDR. [Jul 6 15:53:11] DEBUG[2909] channel.c: Not copying variable SIPREFERRINGCONTEXT. [Jul 6 15:53:11] DEBUG[2909] channel.c: Not copying variable BRIDGEPVTCALLID. [Jul 6 15:53:11] DEBUG[2909] channel.c: Not copying variable BRIDGEPEER. [Jul 6 15:53:11] DEBUG[2909] channel.c: Not copying variable SIPCALLID. [Jul 6 15:53:11] DEBUG[2909] channel.c: Not copying variable SIPURI. [Jul 6 15:53:11] DEBUG[2909] chan_sip.c: Call for 33i transfered by 0004f215aabb-1@10.0.5.191 [Jul 6 15:53:11] DEBUG[2909] chan_sip.c: Outgoing Call for 33i [Jul 6 15:53:11] DEBUG[2909] chan_sip.c: Updating call counter for outgoing call [Jul 6 15:53:11] DEBUG[2909] chan_sip.c: ** Our capability: 0x4 (ulaw) Video flag: False Text flag: False [Jul 6 15:53:11] DEBUG[2909] chan_sip.c: ** Our prefcodec: 0x4 (ulaw) [Jul 6 15:53:11] VERBOSE[2909] chan_sip.c: Audio is at 10.0.5.191 port 18242 [Jul 6 15:53:11] VERBOSE[2909] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [Jul 6 15:53:11] VERBOSE[2909] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Jul 6 15:53:11] DEBUG[2909] chan_sip.c: -- Done with adding codecs to SDP [Jul 6 15:53:11] DEBUG[2909] chan_sip.c: Done building SDP. Settling with this capability: 0x4 (ulaw) [Jul 6 15:53:11] DEBUG[2909] chan_sip.c: Initializing initreq for method INVITE - callid 529d9e7046df2e2b560b669d67ac25f8@10.0.5.191 [Jul 6 15:53:11] DEBUG[2909] chan_sip.c: Header 0 [ 52]: INVITE sip:33i@10.0.5.196:5060;transport=udp SIP/2.0 [Jul 6 15:53:11] DEBUG[2909] chan_sip.c: Header 1 [ 61]: Via: SIP/2.0/UDP 10.0.5.191:5060;branch=z9hG4bK35f371a6;rport [Jul 6 15:53:11] DEBUG[2909] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jul 6 15:53:11] DEBUG[2909] chan_sip.c: Header 3 [ 47]: From: "33i" ;tag=as4c8066aa [Jul 6 15:53:11] DEBUG[2909] chan_sip.c: Header 4 [ 43]: To: [Jul 6 15:53:11] DEBUG[2909] chan_sip.c: Header 5 [ 29]: Contact: [Jul 6 15:53:11] DEBUG[2909] chan_sip.c: Header 6 [ 52]: Call-ID: 529d9e7046df2e2b560b669d67ac25f8@10.0.5.191 [Jul 6 15:53:11] DEBUG[2909] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Jul 6 15:53:11] DEBUG[2909] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 1.6.2.9 [Jul 6 15:53:11] DEBUG[2909] chan_sip.c: Header 9 [ 35]: Date: Tue, 06 Jul 2010 19:53:11 GMT [Jul 6 15:53:11] DEBUG[2909] chan_sip.c: Header 10 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Jul 6 15:53:11] DEBUG[2909] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Jul 6 15:53:11] DEBUG[2909] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Jul 6 15:53:11] DEBUG[2909] chan_sip.c: Header 13 [ 19]: Content-Length: 232 [Jul 6 15:53:11] DEBUG[2909] chan_sip.c: Header 14 [ 0]: [Jul 6 15:53:11] DEBUG[2909] chan_sip.c: Body 0 [ 3]: v=0 [Jul 6 15:53:11] DEBUG[2909] chan_sip.c: Body 1 [ 46]: o=root 1225390108 1225390108 IN IP4 10.0.5.191 [Jul 6 15:53:11] DEBUG[2909] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 1.6.2.9 [Jul 6 15:53:11] DEBUG[2909] chan_sip.c: Body 3 [ 19]: c=IN IP4 10.0.5.191 [Jul 6 15:53:11] DEBUG[2909] chan_sip.c: Body 4 [ 5]: t=0 0 [Jul 6 15:53:11] DEBUG[2909] chan_sip.c: Body 5 [ 27]: m=audio 18242 RTP/AVP 0 101 [Jul 6 15:53:11] DEBUG[2909] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000 [Jul 6 15:53:11] DEBUG[2909] chan_sip.c: Body 7 [ 33]: a=rtpmap:101 telephone-event/8000 [Jul 6 15:53:11] DEBUG[2909] chan_sip.c: Body 8 [ 15]: a=fmtp:101 0-16 [Jul 6 15:53:11] DEBUG[2909] chan_sip.c: Body 9 [ 10]: a=ptime:20 [Jul 6 15:53:11] DEBUG[2909] chan_sip.c: Body 10 [ 10]: a=sendrecv [Jul 6 15:53:11] VERBOSE[2909] chan_sip.c: Reliably Transmitting (no NAT) to 10.0.5.196:5060: INVITE sip:33i@10.0.5.196:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.0.5.191:5060;branch=z9hG4bK35f371a6;rport Max-Forwards: 70 From: "33i" ;tag=as4c8066aa To: Contact: Call-ID: 529d9e7046df2e2b560b669d67ac25f8@10.0.5.191 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.2.9 Date: Tue, 06 Jul 2010 19:53:11 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 232 v=0 o=root 1225390108 1225390108 IN IP4 10.0.5.191 s=Asterisk PBX 1.6.2.9 c=IN IP4 10.0.5.191 t=0 0 m=audio 18242 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- [Jul 6 15:53:11] DEBUG[2909] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #90 [Jul 6 15:53:11] DEBUG[2909] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 10.0.5.196:5060 [Jul 6 15:53:11] VERBOSE[2909] app_dial.c: -- Called 33i [Jul 6 15:53:11] VERBOSE[2782] chan_sip.c: <--- SIP read from UDP:10.0.5.198:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.5.191:5060;branch=z9hG4bK0bc2b3dc;rport From: ;tag=as545536d7 To: "0004f215aabb-1" ;tag=B04C9344-7C4E7635 CSeq: 104 NOTIFY Call-ID: ab3922f0-ae400522-a64ac11b@10.0.5.198 Contact: Event: presence User-Agent: PolycomSoundPointIP-SPIP_501-UA/3.1.6.0017 Accept-Language: fr-fr,fr;q=0.9,en;q=0.8 Content-Length: 0 <-------------> [Jul 6 15:53:11] DEBUG[2782] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Jul 6 15:53:11] DEBUG[2782] chan_sip.c: Header 1 [ 61]: Via: SIP/2.0/UDP 10.0.5.191:5060;branch=z9hG4bK0bc2b3dc;rport [Jul 6 15:53:11] DEBUG[2782] chan_sip.c: Header 2 [ 41]: From: ;tag=as545536d7 [Jul 6 15:53:11] DEBUG[2782] chan_sip.c: Header 3 [ 74]: To: "0004f215aabb-1" ;tag=B04C9344-7C4E7635 [Jul 6 15:53:11] DEBUG[2782] chan_sip.c: Header 4 [ 16]: CSeq: 104 NOTIFY [Jul 6 15:53:11] DEBUG[2782] chan_sip.c: Header 5 [ 46]: Call-ID: ab3922f0-ae400522-a64ac11b@10.0.5.198 [Jul 6 15:53:11] DEBUG[2782] chan_sip.c: Header 6 [ 40]: Contact: [Jul 6 15:53:11] DEBUG[2782] chan_sip.c: Header 7 [ 15]: Event: presence [Jul 6 15:53:11] DEBUG[2782] chan_sip.c: Header 8 [ 54]: User-Agent: PolycomSoundPointIP-SPIP_501-UA/3.1.6.0017 [Jul 6 15:53:11] DEBUG[2782] chan_sip.c: Header 9 [ 40]: Accept-Language: fr-fr,fr;q=0.9,en;q=0.8 [Jul 6 15:53:11] DEBUG[2782] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Jul 6 15:53:11] DEBUG[2782] chan_sip.c: Header 11 [ 0]: [Jul 6 15:53:11] VERBOSE[2782] chan_sip.c: --- (11 headers 0 lines) --- [Jul 6 15:53:11] DEBUG[2782] chan_sip.c: Acked pending invite 104 [Jul 6 15:53:11] DEBUG[2782] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #89 [Jul 6 15:53:11] DEBUG[2782] chan_sip.c: Stopping retransmission on 'ab3922f0-ae400522-a64ac11b@10.0.5.198' of Request 104: Match Found [Jul 6 15:53:11] VERBOSE[2782] chan_sip.c: SIP Response message for INCOMING dialog NOTIFY arrived [Jul 6 15:53:11] VERBOSE[2782] chan_sip.c: <--- SIP read from UDP:10.0.5.196:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.0.5.191:5060;branch=z9hG4bK35f371a6;rport=5060;received=10.0.5.191 From: "33i" ;tag=as4c8066aa To: ;tag=2616030186 Call-ID: 529d9e7046df2e2b560b669d67ac25f8@10.0.5.191 CSeq: 102 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Contact: "33i" ;+sip.instance="" Server: Aastra 9143i/2.5.3.2002 Supported: gruu, path Content-Length: 0 <-------------> [Jul 6 15:53:11] DEBUG[2782] chan_sip.c: Header 0 [ 19]: SIP/2.0 180 Ringing [Jul 6 15:53:11] DEBUG[2782] chan_sip.c: Header 1 [ 86]: Via: SIP/2.0/UDP 10.0.5.191:5060;branch=z9hG4bK35f371a6;rport=5060;received=10.0.5.191 [Jul 6 15:53:11] DEBUG[2782] chan_sip.c: Header 2 [ 47]: From: "33i" ;tag=as4c8066aa [Jul 6 15:53:11] DEBUG[2782] chan_sip.c: Header 3 [ 58]: To: ;tag=2616030186 [Jul 6 15:53:11] DEBUG[2782] chan_sip.c: Header 4 [ 52]: Call-ID: 529d9e7046df2e2b560b669d67ac25f8@10.0.5.191 [Jul 6 15:53:11] DEBUG[2782] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Jul 6 15:53:11] DEBUG[2782] chan_sip.c: Header 6 [ 88]: Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO [Jul 6 15:53:11] DEBUG[2782] chan_sip.c: Header 7 [ 53]: Allow-Events: talk, hold, conference, LocalModeStatus [Jul 6 15:53:11] DEBUG[2782] chan_sip.c: Header 8 [118]: Contact: "33i" ;+sip.instance="" [Jul 6 15:53:11] DEBUG[2782] chan_sip.c: Header 9 [ 31]: Server: Aastra 9143i/2.5.3.2002 [Jul 6 15:53:11] DEBUG[2782] chan_sip.c: Header 10 [ 21]: Supported: gruu, path [Jul 6 15:53:11] DEBUG[2782] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Jul 6 15:53:11] DEBUG[2782] chan_sip.c: Header 12 [ 0]: [Jul 6 15:53:11] VERBOSE[2782] chan_sip.c: --- (12 headers 0 lines) --- [Jul 6 15:53:11] DEBUG[2782] chan_sip.c: *** SIP TIMER: Cancelling retransmission #90 - INVITE (got response) [Jul 6 15:53:11] DEBUG[2782] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '529d9e7046df2e2b560b669d67ac25f8@10.0.5.191' Request 102: Found [Jul 6 15:53:11] DEBUG[2782] chan_sip.c: SIP response 180 to standard invite [Jul 6 15:53:11] VERBOSE[2909] app_dial.c: -- SIP/33i-00000002 is ringing [Jul 6 15:53:11] DEBUG[2689] devicestate.c: No provider found, checking channel drivers for SIP - 33i [Jul 6 15:53:11] DEBUG[2689] chan_sip.c: Checking device state for peer 33i [Jul 6 15:53:11] DEBUG[2689] devicestate.c: Changing state for SIP/33i - state 1 (Not in use) [Jul 6 15:53:11] DEBUG[2689] devicestate.c: device 'SIP/33i' state '1' [Jul 6 15:53:11] DEBUG[2909] rtp.c: Setting early bridge SDP of 'SIP/33i-00000000' with that of 'SIP/33i-00000002' [Jul 6 15:53:11] DEBUG[2909] channel.c: Driver for channel 'SIP/33i-00000000' does not support indication 3, emulating it [Jul 6 15:53:11] DEBUG[2909] channel.c: Set channel SIP/33i-00000000 to write format slin [Jul 6 15:53:11] DEBUG[2909] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [Jul 6 15:53:11] DEBUG[2779] app_queue.c: Device 'SIP/33i' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Jul 6 15:53:11] DEBUG[2909] rtp.c: Difference is 1984, ms is 268 [Jul 6 15:53:14] VERBOSE[2782] chan_sip.c: <--- SIP read from UDP:10.0.5.196:5060 ---> SIP/2.0 486 Busy Here Via: SIP/2.0/UDP 10.0.5.191:5060;branch=z9hG4bK35f371a6;rport=5060;received=10.0.5.191 From: "33i" ;tag=as4c8066aa To: ;tag=2616030186 Call-ID: 529d9e7046df2e2b560b669d67ac25f8@10.0.5.191 CSeq: 102 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Server: Aastra 9143i/2.5.3.2002 Supported: gruu, path Content-Length: 0 <-------------> [Jul 6 15:53:14] DEBUG[2782] chan_sip.c: Header 0 [ 21]: SIP/2.0 486 Busy Here [Jul 6 15:53:14] DEBUG[2782] chan_sip.c: Header 1 [ 86]: Via: SIP/2.0/UDP 10.0.5.191:5060;branch=z9hG4bK35f371a6;rport=5060;received=10.0.5.191 [Jul 6 15:53:14] DEBUG[2782] chan_sip.c: Header 2 [ 47]: From: "33i" ;tag=as4c8066aa [Jul 6 15:53:14] DEBUG[2782] chan_sip.c: Header 3 [ 58]: To: ;tag=2616030186 [Jul 6 15:53:14] DEBUG[2782] chan_sip.c: Header 4 [ 52]: Call-ID: 529d9e7046df2e2b560b669d67ac25f8@10.0.5.191 [Jul 6 15:53:14] DEBUG[2782] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Jul 6 15:53:14] DEBUG[2782] chan_sip.c: Header 6 [ 88]: Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO [Jul 6 15:53:14] DEBUG[2782] chan_sip.c: Header 7 [ 31]: Server: Aastra 9143i/2.5.3.2002 [Jul 6 15:53:14] DEBUG[2782] chan_sip.c: Header 8 [ 21]: Supported: gruu, path [Jul 6 15:53:14] DEBUG[2782] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Jul 6 15:53:14] DEBUG[2782] chan_sip.c: Header 10 [ 0]: [Jul 6 15:53:14] VERBOSE[2782] chan_sip.c: --- (10 headers 0 lines) --- [Jul 6 15:53:14] DEBUG[2782] chan_sip.c: Acked pending invite 102 [Jul 6 15:53:14] DEBUG[2782] chan_sip.c: Stopping retransmission on '529d9e7046df2e2b560b669d67ac25f8@10.0.5.191' of Request 102: Match Found [Jul 6 15:53:14] VERBOSE[2782] chan_sip.c: -- Got SIP response 486 "Busy Here" back from 10.0.5.196 [Jul 6 15:53:14] VERBOSE[2782] chan_sip.c: Transmitting (no NAT) to 10.0.5.196:5060: ACK sip:33i@10.0.5.196:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.0.5.191:5060;branch=z9hG4bK35f371a6;rport Max-Forwards: 70 From: "33i" ;tag=as4c8066aa To: ;tag=2616030186 Contact: Call-ID: 529d9e7046df2e2b560b669d67ac25f8@10.0.5.191 CSeq: 102 ACK User-Agent: Asterisk PBX 1.6.2.9 Content-Length: 0 --- [Jul 6 15:53:14] DEBUG[2782] chan_sip.c: Trying to put 'ACK sip:33i' onto UDP socket destined for 10.0.5.196:5060 [Jul 6 15:53:14] DEBUG[2782] chan_sip.c: Setting SIP_ALREADYGONE on dialog 529d9e7046df2e2b560b669d67ac25f8@10.0.5.191 [Jul 6 15:53:14] VERBOSE[2909] app_dial.c: -- SIP/33i-00000002 is busy [Jul 6 15:53:14] DEBUG[2909] channel.c: Hanging up channel 'SIP/33i-00000002' [Jul 6 15:53:14] DEBUG[2909] chan_sip.c: Hangup call SIP/33i-00000002, SIP callid 529d9e7046df2e2b560b669d67ac25f8@10.0.5.191 [Jul 6 15:53:14] VERBOSE[2909] app_dial.c: == Everyone is busy/congested at this time (1:1/0/0) [Jul 6 15:53:14] DEBUG[2909] rtp.c: Channel '' has no RTP, not doing anything [Jul 6 15:53:14] DEBUG[2909] app_dial.c: Exiting with DIALSTATUS=BUSY. [Jul 6 15:53:14] VERBOSE[2909] pbx.c: -- Auto fallthrough, channel 'SIP/33i-00000000' status is 'BUSY' [Jul 6 15:53:14] DEBUG[2909] channel.c: Driver for channel 'SIP/33i-00000000' does not support indication 5, emulating it [Jul 6 15:53:14] DEBUG[2909] channel.c: Set channel SIP/33i-00000000 to write format ulaw [Jul 6 15:53:14] DEBUG[2909] channel.c: Set channel SIP/33i-00000000 to write format slin [Jul 6 15:53:14] DEBUG[2909] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [Jul 6 15:53:14] DEBUG[2689] devicestate.c: No provider found, checking channel drivers for SIP - 33i [Jul 6 15:53:14] DEBUG[2689] chan_sip.c: Checking device state for peer 33i [Jul 6 15:53:14] DEBUG[2689] devicestate.c: Changing state for SIP/33i - state 1 (Not in use) [Jul 6 15:53:14] DEBUG[2689] devicestate.c: device 'SIP/33i' state '1' [Jul 6 15:53:14] DEBUG[2779] app_queue.c: Device 'SIP/33i' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Jul 6 15:53:14] DEBUG[2909] rtp.c: Got RTCP report of 64 bytes [Jul 6 15:53:15] DEBUG[2782] chan_sip.c: Destroying SIP dialog 529d9e7046df2e2b560b669d67ac25f8@10.0.5.191 [Jul 6 15:53:15] VERBOSE[2782] chan_sip.c: Really destroying SIP dialog '529d9e7046df2e2b560b669d67ac25f8@10.0.5.191' Method: INVITE [Jul 6 15:53:18] VERBOSE[2782] chan_sip.c: <--- SIP read from UDP:10.0.5.196:5060 ---> BYE sip:901@10.0.5.191 SIP/2.0 Via: SIP/2.0/UDP 10.0.5.196:5060;branch=z9hG4bK097ec5a99da1842e8.1de6e5106b6a26cd3 Max-Forwards: 70 From: "33i" ;tag=8497c4b8bd To: "901" ;tag=as6ef584fe Call-ID: f7bf38449b305573 CSeq: 9033 BYE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Authorization: Digest username="33i",realm="asterisk",nonce="05793f28",uri="sip:901@10.0.5.191",response="60d3464f0f4392a5cf33f75665082134",algorithm=MD5 Supported: gruu, path, timer User-Agent: Aastra 9143i/2.5.3.2002 Content-Length: 0 <-------------> [Jul 6 15:53:18] DEBUG[2782] chan_sip.c: Header 0 [ 30]: BYE sip:901@10.0.5.191 SIP/2.0 [Jul 6 15:53:18] DEBUG[2782] chan_sip.c: Header 1 [ 82]: Via: SIP/2.0/UDP 10.0.5.196:5060;branch=z9hG4bK097ec5a99da1842e8.1de6e5106b6a26cd3 [Jul 6 15:53:18] DEBUG[2782] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jul 6 15:53:18] DEBUG[2782] chan_sip.c: Header 3 [ 52]: From: "33i" ;tag=8497c4b8bd [Jul 6 15:53:18] DEBUG[2782] chan_sip.c: Header 4 [ 50]: To: "901" ;tag=as6ef584fe [Jul 6 15:53:18] DEBUG[2782] chan_sip.c: Header 5 [ 25]: Call-ID: f7bf38449b305573 [Jul 6 15:53:18] DEBUG[2782] chan_sip.c: Header 6 [ 14]: CSeq: 9033 BYE [Jul 6 15:53:18] DEBUG[2782] chan_sip.c: Header 7 [ 88]: Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO [Jul 6 15:53:18] DEBUG[2782] chan_sip.c: Header 8 [ 53]: Allow-Events: talk, hold, conference, LocalModeStatus [Jul 6 15:53:18] DEBUG[2782] chan_sip.c: Header 9 [153]: Authorization: Digest username="33i",realm="asterisk",nonce="05793f28",uri="sip:901@10.0.5.191",response="60d3464f0f4392a5cf33f75665082134",algorithm=MD5 [Jul 6 15:53:18] DEBUG[2782] chan_sip.c: Header 10 [ 28]: Supported: gruu, path, timer [Jul 6 15:53:18] DEBUG[2782] chan_sip.c: Header 11 [ 35]: User-Agent: Aastra 9143i/2.5.3.2002 [Jul 6 15:53:18] DEBUG[2782] chan_sip.c: Header 12 [ 17]: Content-Length: 0 [Jul 6 15:53:18] DEBUG[2782] chan_sip.c: Header 13 [ 0]: [Jul 6 15:53:18] VERBOSE[2782] chan_sip.c: --- (13 headers 0 lines) --- [Jul 6 15:53:18] DEBUG[2782] chan_sip.c: **** Received BYE (8) - Command in SIP BYE [Jul 6 15:53:18] DEBUG[2782] chan_sip.c: Initializing initreq for method BYE - callid f7bf38449b305573 [Jul 6 15:53:18] VERBOSE[2782] chan_sip.c: Sending to 10.0.5.196 : 5060 (no NAT) [Jul 6 15:53:18] DEBUG[2782] chan_sip.c: Setting SIP_ALREADYGONE on dialog f7bf38449b305573 [Jul 6 15:53:18] DEBUG[2782] chan_sip.c: Session timer stopped: -1 - f7bf38449b305573 [Jul 6 15:53:18] DEBUG[2782] chan_sip.c: Received bye, issuing owner hangup [Jul 6 15:53:18] VERBOSE[2782] chan_sip.c: <--- Transmitting (no NAT) to 10.0.5.196:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.5.196:5060;branch=z9hG4bK097ec5a99da1842e8.1de6e5106b6a26cd3;received=10.0.5.196 From: "33i" ;tag=8497c4b8bd To: "901" ;tag=as6ef584fe Call-ID: f7bf38449b305573 CSeq: 9033 BYE Server: Asterisk PBX 1.6.2.9 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 <------------> [Jul 6 15:53:18] DEBUG[2782] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.0.5.196:5060 [Jul 6 15:53:18] DEBUG[2909] channel.c: Set channel SIP/33i-00000000 to write format ulaw [Jul 6 15:53:18] DEBUG[2909] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Jul 6 15:53:18] DEBUG[2909] channel.c: Soft-Hanging up channel 'SIP/33i-00000000' [Jul 6 15:53:18] DEBUG[2909] channel.c: Hanging up channel 'SIP/33i-00000000' [Jul 6 15:53:18] DEBUG[2909] chan_sip.c: Hangup call SIP/33i-00000000, SIP callid f7bf38449b305573 [Jul 6 15:53:18] DEBUG[2689] devicestate.c: No provider found, checking channel drivers for SIP - 33i [Jul 6 15:53:18] DEBUG[2689] chan_sip.c: Checking device state for peer 33i [Jul 6 15:53:18] DEBUG[2689] devicestate.c: Changing state for SIP/33i - state 1 (Not in use) [Jul 6 15:53:18] DEBUG[2689] devicestate.c: device 'SIP/33i' state '1' [Jul 6 15:53:18] DEBUG[2779] app_queue.c: Device 'SIP/33i' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Jul 6 15:53:19] DEBUG[2782] chan_sip.c: Destroying SIP dialog f7bf38449b305573 [Jul 6 15:53:19] VERBOSE[2782] chan_sip.c: Really destroying SIP dialog 'f7bf38449b305573' Method: BYE [Jul 6 15:53:22] DEBUG[2782] chan_sip.c: Auto destroying SIP dialog 'c8ce19c6966b23d9' [Jul 6 15:53:22] DEBUG[2782] chan_sip.c: Destroying SIP dialog c8ce19c6966b23d9 [Jul 6 15:53:22] VERBOSE[2782] chan_sip.c: Really destroying SIP dialog 'c8ce19c6966b23d9' Method: REGISTER [Jul 6 15:53:35] DEBUG[2782] acl.c: Found IP address for this socket [Jul 6 15:53:35] DEBUG[2782] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 10.0.5.191:5060 [Jul 6 15:53:35] DEBUG[2782] chan_sip.c: Allocating new SIP dialog for c8ce19c6966b23d9 - REGISTER (No RTP) [Jul 6 15:53:35] DEBUG[2782] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Jul 6 15:53:35] DEBUG[2782] chan_sip.c: Trying to put 'SIP/2.0 401' onto UDP socket destined for 10.0.5.196:5060 [Jul 6 15:53:35] DEBUG[2782] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Jul 6 15:53:35] DEBUG[2782] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.0.5.196:5060 [Jul 6 15:53:35] DEBUG[2689] devicestate.c: No provider found, checking channel drivers for SIP - 33i [Jul 6 15:53:35] DEBUG[2689] chan_sip.c: Checking device state for peer 33i [Jul 6 15:53:35] DEBUG[2689] devicestate.c: Changing state for SIP/33i - state 1 (Not in use) [Jul 6 15:53:35] DEBUG[2689] devicestate.c: device 'SIP/33i' state '1' [Jul 6 15:53:35] DEBUG[2779] app_queue.c: Device 'SIP/33i' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Jul 6 15:53:35] DEBUG[2782] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Jul 6 15:53:35] DEBUG[2782] chan_sip.c: Trying to put 'SIP/2.0 401' onto UDP socket destined for 10.0.5.198:5060 [Jul 6 15:53:35] DEBUG[2782] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Jul 6 15:53:35] DEBUG[2782] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.0.5.198:5060 [Jul 6 15:53:35] DEBUG[2689] devicestate.c: No provider found, checking channel drivers for SIP - 0004f215aabb-1 [Jul 6 15:53:35] DEBUG[2689] chan_sip.c: Checking device state for peer 0004f215aabb-1 [Jul 6 15:53:35] DEBUG[2689] devicestate.c: Changing state for SIP/0004f215aabb-1 - state 1 (Not in use) [Jul 6 15:53:35] DEBUG[2689] devicestate.c: device 'SIP/0004f215aabb-1' state '1' [Jul 6 15:53:35] DEBUG[2779] app_queue.c: Device 'SIP/0004f215aabb-1' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Jul 6 15:53:36] DEBUG[2782] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Jul 6 15:53:36] DEBUG[2782] chan_sip.c: Trying to put 'SIP/2.0 401' onto UDP socket destined for 10.0.5.198:5060 [Jul 6 15:53:36] DEBUG[2782] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Jul 6 15:53:36] DEBUG[2782] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.0.5.198:5060 [Jul 6 15:53:36] DEBUG[2689] devicestate.c: No provider found, checking channel drivers for SIP - 0004f215aabb-2 [Jul 6 15:53:36] DEBUG[2689] chan_sip.c: Checking device state for peer 0004f215aabb-2 [Jul 6 15:53:36] DEBUG[2689] devicestate.c: Changing state for SIP/0004f215aabb-2 - state 1 (Not in use) [Jul 6 15:53:36] DEBUG[2689] devicestate.c: device 'SIP/0004f215aabb-2' state '1' [Jul 6 15:53:36] DEBUG[2779] app_queue.c: Device 'SIP/0004f215aabb-2' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.