<<< Log from 10.1.1.1 started Oktober 04, 2004, 16:51:35 >>> sip debug SIP Debugging Enabled *CLI> Sip read: INVITE sip:3249211@10.1.1.1 SIP/2.0 Via: SIP/2.0/UDP 10.1.41.188;branch=z9hG4bKccf2df6e460e1488 From: ;tag=07173f82c8e9b3ab To: Contact: Call-ID: 94c4665f63bf3f3a@10.1.41.188 CSeq: 61902 INVITE User-Agent: Grandstream BT100 1.0.5.11 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Type: application/sdp Content-Length: 337 v=0 o=GSIn 8000 8000 IN IP4 10.1.41.188 s=SIP Call c=IN IP4 10.1.41.188 t=0 0 m=audio 5004 RTP/AVP 99 2 0 8 18 15 9 4 a=rtpmap:99 iLBC/8000 a=fmtp:99 mode=20 a=rtpmap:2 G726-32/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:15 G728/8000 a=rtpmap:9 G722/8000 a=rtpmap:4 G723/8000 a=ptime:40 12 headers, 16 lines Using latest request as basis request Urgent handler Sending to 10.1.41.188 : 5060 (non-NAT) Urgent handler Found RTP audio format 99 Found RTP audio format 2 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 18 Found RTP audio format 15 Found RTP audio format 9 Found RTP audio format 4 Urgent handler Peer audio RTP is at port 10.1.41.188:5004 Found description format iLBC Found description format G726-32 Found description format PCMU Found description format PCMA Found description format G729 Found description format G728 Urgent handler Found description format G722 Found description format G723 Capabilities: us - 0x8000e(GSM|ULAW|ALAW|H263), peer - audio=0x51d(G723|ULAW|ALA W|G726|G729A|ILBC)/video=0x0(EMPTY), combined - 0xc(ULAW|ALAW) Non-codec capabilities: us - 0x1(G723), peer - 0x0(EMPTY), combined - 0x0(EMPTY) Reliably Transmitting (no NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 10.1.41.188;branch=z9hG4bKccf2df6e460e1488 From: ;tag=07173f82c8e9b3ab To: ;tag=as587f02b6 Call-ID: 94c4665f63bf3f3a@10.1.41.188 CSeq: 61902 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Proxy-Authenticate: Digest realm="asterisk", nonce="34061f82" Content-Length: 0 to 10.1.41.188:5060 Scheduling destruction of call '94c4665f63bf3f3a@10.1.41.188' in 15000 ms Found user 'GSIn' Urgent handler Urgent handler Sip read: ACK sip:3249211@10.1.1.1 SIP/2.0 Via: SIP/2.0/UDP 10.1.41.188;branch=z9hG4bKccf2df6e460e1488 From: ;tag=07173f82c8e9b3ab To: ;tag=as587f02b6 Contact: Call-ID: 94c4665f63bf3f3a@10.1.41.188 CSeq: 61902 ACK User-Agent: Grandstream BT100 1.0.5.11 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Length: 0 11 headers, 0 lines Urgent handler Sip read: INVITE sip:3249211@10.1.1.1 SIP/2.0 Via: SIP/2.0/UDP 10.1.41.188;branch=z9hG4bK17aa4d25a4af4628 From: ;tag=07173f82c8e9b3ab To: Contact: Proxy-Authorization: DIGEST username="GSIn", realm="asterisk", algorithm=MD5, ur i="sip:3249211@10.1.1.1", nonce="34061f82", response="f7b52cfbd1a8c355686a536696 ecbb03" Call-ID: 94c4665f63bf3f3a@10.1.41.188 CSeq: 61903 INVITE User-Agent: Grandstream BT100 1.0.5.11 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Type: application/sdp Content-Length: 337 v=0 o=GSIn 8000 8000 IN IP4 10.1.41.188 s=SIP Call c=IN IP4 10.1.41.188 t=0 0 m=audio 5004 RTP/AVP 99 2 0 8 18 15 9 4 a=rtpmap:99 iLBC/8000 a=fmtp:99 mode=20 a=rtpmap:2 G726-32/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:15 G728/8000 a=rtpmap:9 G722/8000 a=rtpmap:4 G723/8000 a=ptime:40 13 headers, 16 lines Using latest request as basis request Sending to 10.1.41.188 : 5060 (non-NAT) Found RTP audio format 99 Found RTP audio format 2 Urgent handler Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 18 Found RTP audio format 15 Found RTP audio format 9 Found RTP audio format 4 Peer audio RTP is at port 10.1.41.188:5004 Found description format iLBC Urgent handler Found description format G726-32 Found description format PCMU Found description format PCMA Found description format G729 Found description format G728 Found description format G722 Found description format G723 Capabilities: us - 0x8000e(GSM|ULAW|ALAW|H263), peer - audio=0x51d(G723|ULAW|ALA W|G726|G729A|ILBC)/video=0x0(EMPTY), combined - 0xc(ULAW|ALAW) Urgent handler Non-codec capabilities: us - 0x1(G723), peer - 0x0(EMPTY), combined - 0x0(EMPTY) Found user 'GSIn' Looking for 3249211 in intern Urgent handler list_route: hop: Transmitting (no NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.1.41.188;branch=z9hG4bK17aa4d25a4af4628 From: ;tag=07173f82c8e9b3ab To: ;tag=as7236abbe Call-ID: 94c4665f63bf3f3a@10.1.41.188 CSeq: 61903 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 to 10.1.41.188:5060 Urgent handler Urgent handler We're at 217.162.36.14 port 17010 Urgent handler Answering with capability 0x400(ILBC) Urgent handler 12 headers, 8 lines Urgent handler Reliably Transmitting: INVITE sip:0041313249211@sipgate.de SIP/2.0 Via: SIP/2.0/UDP 217.162.36.14:5060;branch=z9hG4bK3d2c87f5 From: "1838074" ;tag=as4271ea69 To: Contact: Call-ID: 799800825c74fbce4c983a2022a14199@sipgate.de CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Mon, 04 Oct 2004 14:51:46 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 164 v=0 o=root 20506 20506 IN IP4 217.162.36.14 s=session c=IN IP4 217.162.36.14 t=0 0 m=audio 17010 RTP/AVP 97 a=rtpmap:97 iLBC/8000 a=silenceSupp:off - - - - (no NAT) to 217.10.79.9:5060 Urgent handler Sip read: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 217.162.36.14:5060;branch=z9hG4bK3d2c87f5 From: "1838074" ;tag=as4271ea69 To: ;tag=b11cb9bb270104b49a99a995b8c68544.3c13 Call-ID: 799800825c74fbce4c983a2022a14199@sipgate.de CSeq: 102 INVITE Proxy-Authenticate: Digest realm="sipgate.de", nonce="416164b0b1f22ef2061569c40d fc96e2bc97701a" Server: sipgate ser Content-Length: 0 Warning: 392 217.10.79.9:5060 "Noisy feedback tells: pid=783 req_src_ip=217.162 .36.14 req_src_port=5060 in_uri=sip:0041313249211@sipgate.de out_uri=sip:0041313 249211@sipgate.de via_cnt==1" 10 headers, 0 lines Urgent handler Transmitting: ACK sip:0041313249211@sipgate.de SIP/2.0 Via: SIP/2.0/UDP 217.162.36.14:5060;branch=z9hG4bK3d2c87f5 From: "1838074" ;tag=as4271ea69 To: ;tag=b11cb9bb270104b49a99a995b8c68544.3c13 Contact: Call-ID: 799800825c74fbce4c983a2022a14199@sipgate.de CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 (no NAT) to 217.10.79.9:5060 We're at 217.162.36.14 port 17010 Answering with capability 0x400(ILBC) Urgent handler Reliably Transmitting: INVITE sip:0041313249211@sipgate.de SIP/2.0 Via: SIP/2.0/UDP 217.162.36.14:5060;branch=z9hG4bK5650bd13 From: "1838074" ;tag=as4271ea69 To: Contact: Call-ID: 799800825c74fbce4c983a2022a14199@sipgate.de CSeq: 103 INVITE User-Agent: Asterisk PBX Proxy-Authorization: Digest username="1838074", realm="sipgate.de", algorithm=MD 5, uri="sip:0041313249211@sipgate.de", nonce="416164b0b1f22ef2061569c40dfc96e2bc 97701a", response="1bd3b2086a8cab666989ce5415a53ff1", opaque="" Date: Mon, 04 Oct 2004 14:51:46 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 164 v=0 o=root 20506 20507 IN IP4 217.162.36.14 s=session c=IN IP4 217.162.36.14 t=0 0 m=audio 17010 RTP/AVP 97 a=rtpmap:97 iLBC/8000 a=silenceSupp:off - - - - (no NAT) to 217.10.79.9:5060 Urgent handler Sip read: SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 217.162.36.14:5060;branch=z9hG4bK5650bd13 From: "1838074" ;tag=as4271ea69 To: Call-ID: 799800825c74fbce4c983a2022a14199@sipgate.de CSeq: 103 INVITE Server: sipgate ser Content-Length: 0 Warning: 392 217.10.79.9:5060 "Noisy feedback tells: pid=785 req_src_ip=217.162 .36.14 req_src_port=5060 in_uri=sip:0041313249211@sipgate.de out_uri=sip:4131324 9211@sipgate.net via_cnt==1" 9 headers, 0 lines Urgent handler Sip read: SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 217.162.36.14:5060;branch=z9hG4bK5650bd13 From: "1838074" ;tag=as4271ea69 To: ;tag=as5cfb82e7 Call-ID: 799800825c74fbce4c983a2022a14199@sipgate.de CSeq: 103 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Type: application/sdp Content-Length: 162 v=0 o=root 21451 21451 IN IP4 217.10.79.30 s=session c=IN IP4 217.10.79.30 t=0 0 m=audio 18986 RTP/AVP 97 a=rtpmap:97 iLBC/8000 a=silenceSupp:off - - - - 11 headers, 8 lines Urgent handler Found RTP audio format 97 Peer audio RTP is at port 217.10.79.30:18986 Found description format iLBC Capabilities: us - 0x400(ILBC), peer - audio=0x400(ILBC)/video=0x0(EMPTY), combi ned - 0x400(ILBC) Non-codec capabilities: us - 0x1(G723), peer - 0x0(EMPTY), combined - 0x0(EMPTY) Urgent handler We're at 10.1.1.1 port 16690 Answering with capability 0x2(GSM) Answering with capability 0x4(ULAW) Answering with capability 0x8(ALAW) Transmitting (no NAT): SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 10.1.41.188;branch=z9hG4bK17aa4d25a4af4628 From: ;tag=07173f82c8e9b3ab To: ;tag=as7236abbe Call-ID: 94c4665f63bf3f3a@10.1.41.188 CSeq: 61903 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Type: application/sdp Content-Length: 199 v=0 o=root 20506 20506 IN IP4 10.1.1.1 s=session c=IN IP4 10.1.1.1 t=0 0 m=audio 16690 RTP/AVP 3 0 8 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=silenceSupp:off - - - - to 10.1.41.188:5060 Urgent handler Urgent handler Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 217.162.36.14:5060;branch=z9hG4bK5650bd13 Record-Route: Record-Route: From: "1838074" ;tag=as4271ea69 To: ;tag=as5cfb82e7 Call-ID: 799800825c74fbce4c983a2022a14199@sipgate.de CSeq: 103 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Type: application/sdp Content-Length: 162 v=0 o=root 21451 21452 IN IP4 217.10.79.30 s=session c=IN IP4 217.10.79.30 t=0 0 m=audio 18986 RTP/AVP 97 a=rtpmap:97 iLBC/8000 a=silenceSupp:off - - - - 13 headers, 8 lines Found RTP audio format 97 Peer audio RTP is at port 217.10.79.30:18986 Found description format iLBC Capabilities: us - 0x400(ILBC), peer - audio=0x400(ILBC)/video=0x0(EMPTY), combi ned - 0x400(ILBC) Non-codec capabilities: us - 0x1(G723), peer - 0x0(EMPTY), combined - 0x0(EMPTY) list_route: hop: list_route: hop: list_route: hop: set_destination: Parsing f or address/port to send to set_destination: set destination to 217.10.79.9, port 5060 Transmitting: ACK sip:0041313249211@sipgate.de SIP/2.0 Via: SIP/2.0/UDP 217.162.36.14:5060;branch=z9hG4bK2f1179ca Route: , From: "1838074" ;tag=as4271ea69 To: ;tag=as5cfb82e7 Contact: Call-ID: 799800825c74fbce4c983a2022a14199@sipgate.de CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (no NAT) to 217.10.79.9:5060 Urgent handler Urgent handler Oct 4 16:51:49 NOTICE[229390]: chan_sip.c:1853 sip_answer: Changing codec to 'i lbc' for this call because of ${SIP_CODEC) variable Oct 4 16:51:49 NOTICE[229390]: chan_sip.c:1858 sip_answer: Ignoring ${SIP_CODEC } variable because it is not shared by both ends. We're at 10.1.1.1 port 16690 Answering with capability 0x2(GSM) Urgent handler Answering with capability 0x4(ULAW) Answering with capability 0x8(ALAW) Reliably Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.41.188;branch=z9hG4bK17aa4d25a4af4628 From: ;tag=07173f82c8e9b3ab To: ;tag=as7236abbe Call-ID: 94c4665f63bf3f3a@10.1.41.188 CSeq: 61903 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Type: application/sdp Content-Length: 199 v=0 o=root 20506 20507 IN IP4 10.1.1.1 s=session c=IN IP4 10.1.1.1 t=0 0 m=audio 16690 RTP/AVP 3 0 8 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=silenceSupp:off - - - - to 10.1.41.188:5060 Urgent handler Sip read: ACK sip:3249211@10.1.1.1 SIP/2.0 Via: SIP/2.0/UDP 10.1.41.188;branch=z9hG4bK882ea01e87f2607d From: ;tag=07173f82c8e9b3ab To: ;tag=as7236abbe Contact: Proxy-Authorization: DIGEST username="GSIn", realm="asterisk", algorithm=MD5, ur i="sip:3249211@10.1.1.1", nonce="34061f82", response="dae66c49c9d483bdd94fbc823d 8e2598" Call-ID: 94c4665f63bf3f3a@10.1.41.188 CSeq: 61903 ACK User-Agent: Grandstream BT100 1.0.5.11 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Length: 0 12 headers, 0 lines Urgent handler sip show channels Peer User/ANR Call ID Seq (Tx/Rx) Format 217.10.79.9 0041313249 799800825c7 00103/00000 ILBC 10.1.41.188 GSIn 94c4665f63b 00101/61903 ULAW 2 active SIP channel(s) *CLI> Sip read: BYE sip:3249211@10.1.1.1 SIP/2.0 Via: SIP/2.0/UDP 10.1.41.188;branch=z9hG4bK5fed2178a7f8f373 From: ;tag=07173f82c8e9b3ab To: ;tag=as7236abbe Contact: Proxy-Authorization: DIGEST username="GSIn", realm="asterisk", algorithm=MD5, ur i="sip:3249211@10.1.1.1", nonce="34061f82", response="24d41adf6d97a6cabc59c2f413 1ef71e" Call-ID: 94c4665f63bf3f3a@10.1.41.188 CSeq: 61904 BYE User-Agent: Grandstream BT100 1.0.5.11 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE From: ;tag=07173f82c8e9b3ab To: ;tag=as7236abbe Call-ID: 94c4665f63bf3f3a@10.1.41.188 CSeq: 61904 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 to 10.1.41.188:5060 To: ;tag=as5cfb82e7 rufus:/usr/lib/asterisk/modules{root}[146]> <<< Log from 10.1.1.1 ended Oktober 04, 2004, 16:52:02 >>>