[Mar 2 16:08:45] VERBOSE[30149] logger.c: <--- SIP read from 11.111.32.132:5061 ---> INVITE sip:001555551636@app.fooobar.at:5160 SIP/2.0 Via: SIP/2.0/UDP 11.111.32.132:5061;branch=z9hG4bK-24391-1-0 From: ;tag=1 To: Call-ID: 1-24391@11.111.32.132 CSeq: 1 INVITE Contact: sip:sipp@11.111.32.132:5061 Max-Forwards: 70 Subject: Tests performed by darilion@ipcom.at Content-Type: application/sdp Content-Length: 135 v=0 o=user1 53655765 2353687637 IN IP4 11.111.32.132 s=- c=IN IP4 11.111.32.132 t=0 0 m=audio 6000 RTP/AVP 0 a=rtpmap:0 PCMU/8000 <-------------> [Mar 2 16:08:45] DEBUG[30149] chan_sip.c: Header 0: INVITE sip:001555551636@app.fooobar.at:5160 SIP/2.0 (51) [Mar 2 16:08:45] DEBUG[30149] chan_sip.c: Header 1: Via: SIP/2.0/UDP 11.111.32.132:5061;branch=z9hG4bK-24391-1-0 (60) [Mar 2 16:08:45] DEBUG[30149] chan_sip.c: Header 2: From: ;tag=1 (50) [Mar 2 16:08:45] DEBUG[30149] chan_sip.c: Header 3: To: (42) [Mar 2 16:08:45] DEBUG[30149] chan_sip.c: Header 4: Call-ID: 1-24391@11.111.32.132 (30) [Mar 2 16:08:45] DEBUG[30149] chan_sip.c: Header 5: CSeq: 1 INVITE (14) [Mar 2 16:08:45] DEBUG[30149] chan_sip.c: Header 6: Contact: sip:sipp@11.111.32.132:5061 (36) [Mar 2 16:08:45] DEBUG[30149] chan_sip.c: Header 7: Max-Forwards: 70 (16) [Mar 2 16:08:45] DEBUG[30149] chan_sip.c: Header 8: Subject: Tests performed by darilion@ipcom.at (45) [Mar 2 16:08:45] DEBUG[30149] chan_sip.c: Header 9: Content-Type: application/sdp (29) [Mar 2 16:08:45] DEBUG[30149] chan_sip.c: Header 10: Content-Length: 135 (19) [Mar 2 16:08:45] DEBUG[30149] chan_sip.c: Header 11: (0) [Mar 2 16:08:45] DEBUG[30149] chan_sip.c: Line: v=0 (3) [Mar 2 16:08:45] DEBUG[30149] chan_sip.c: Line: o=user1 53655765 2353687637 IN IP4 11.111.32.132 (48) [Mar 2 16:08:45] DEBUG[30149] chan_sip.c: Line: s=- (3) [Mar 2 16:08:45] DEBUG[30149] chan_sip.c: Line: c=IN IP4 11.111.32.132 (22) [Mar 2 16:08:45] DEBUG[30149] chan_sip.c: Line: t=0 0 (5) [Mar 2 16:08:45] DEBUG[30149] chan_sip.c: Line: m=audio 6000 RTP/AVP 0 (22) [Mar 2 16:08:45] VERBOSE[30149] logger.c: --- (11 headers 7 lines) --- [Mar 2 16:08:45] DEBUG[30149] chan_sip.c: = Looking for Call ID: 1-24391@11.111.32.132 (Checking From) --From tag 1 --To-tag [Mar 2 16:08:45] DEBUG[30149] chan_sip.c: Setting NAT on RTP to On [Mar 2 16:08:45] DEBUG[30149] chan_sip.c: Setting NAT on UDPTL to On [Mar 2 16:08:45] DEBUG[30149] chan_sip.c: Allocating new SIP dialog for 1-24391@11.111.32.132 - INVITE (With RTP) [Mar 2 16:08:45] DEBUG[30149] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [Mar 2 16:08:45] VERBOSE[30149] logger.c: Sending to 11.111.32.132 : 5061 (NAT) [Mar 2 16:08:45] VERBOSE[30149] logger.c: Using INVITE request as basis request - 1-24391@11.111.32.132 [Mar 2 16:08:45] DEBUG[30149] chan_sip.c: Setting NAT on RTP to On [Mar 2 16:08:45] DEBUG[30149] chan_sip.c: Setting NAT on UDPTL to On [Mar 2 16:08:45] VERBOSE[30149] logger.c: <--- Reliably Transmitting (NAT) to 11.111.32.132:5061 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 11.111.32.132:5061;branch=z9hG4bK-24391-1-0;received=11.111.32.132 From: ;tag=1 To: ;tag=as5f62ade5 Call-ID: 1-24391@11.111.32.132 CSeq: 1 INVITE User-Agent: MegaSIP-app Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm="app.fooobar.at", nonce="5ed78c32" Content-Length: 0 <------------> [Mar 2 16:08:45] DEBUG[30149] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #-1 [Mar 2 16:08:45] VERBOSE[30149] logger.c: Scheduling destruction of SIP dialog '1-24391@11.111.32.132' in 32000 ms (Method: INVITE) [Mar 2 16:08:45] VERBOSE[30149] logger.c: Found user 'u+431234567730151' [Mar 2 16:08:45] VERBOSE[30149] logger.c: <--- SIP read from 11.111.32.132:5061 ---> ACK sip:001555551636@app.fooobar.at:5160 SIP/2.0 Via: SIP/2.0/UDP 11.111.32.132:5061;branch=z9hG4bK-24391-1-0 From: ;tag=1 To: ;tag=as5f62ade5 Call-ID: 1-24391@11.111.32.132 CSeq: 1 ACK Max-Forwards: 70 <-------------> [Mar 2 16:08:45] DEBUG[30149] chan_sip.c: Header 0: ACK sip:001555551636@app.fooobar.at:5160 SIP/2.0 (48) [Mar 2 16:08:45] DEBUG[30149] chan_sip.c: Header 1: Via: SIP/2.0/UDP 11.111.32.132:5061;branch=z9hG4bK-24391-1-0 (60) [Mar 2 16:08:45] DEBUG[30149] chan_sip.c: Header 2: From: ;tag=1 (50) [Mar 2 16:08:45] DEBUG[30149] chan_sip.c: Header 3: To: ;tag=as5f62ade5 (57) [Mar 2 16:08:45] DEBUG[30149] chan_sip.c: Header 4: Call-ID: 1-24391@11.111.32.132 (30) [Mar 2 16:08:45] DEBUG[30149] chan_sip.c: Header 5: CSeq: 1 ACK (11) [Mar 2 16:08:45] DEBUG[30149] chan_sip.c: Header 6: Max-Forwards: 70 (16) [Mar 2 16:08:45] VERBOSE[30149] logger.c: --- (7 headers 0 lines) --- [Mar 2 16:08:45] DEBUG[30149] chan_sip.c: = Looking for Call ID: 1-24391@11.111.32.132 (Checking From) --From tag 1 --To-tag as5f62ade5 [Mar 2 16:08:45] DEBUG[30149] chan_sip.c: = Found Their Call ID: 1-24391@11.111.32.132 Their Tag 1 Our tag: as5f62ade5 [Mar 2 16:08:45] DEBUG[30149] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [Mar 2 16:08:45] DEBUG[30149] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #81137 [Mar 2 16:08:45] DEBUG[30149] chan_sip.c: Stopping retransmission on '1-24391@11.111.32.132' of Response 1: Match Found [Mar 2 16:08:45] VERBOSE[30149] logger.c: <--- SIP read from 11.111.32.132:5061 ---> INVITE sip:001555551636@app.fooobar.at:5160 SIP/2.0 Via: SIP/2.0/UDP 11.111.32.132:5061;branch=z9hG4bK-24391-1-4 From: ;tag=1 To: Proxy-Authorization: Digest username="u+431234567730151",realm="app.fooobar.at",uri="sip:22.22.222.184:5160",nonce="5ed78c32",response="438333757600bacc0aa1ca3f722821b4",algorithm=MD5 Call-ID: 1-24391@11.111.32.132 CSeq: 2 INVITE Contact: sip:sipp@11.111.32.132:5061 Max-Forwards: 70 Subject: Tests performed by darilion@ipcom.at Content-Type: application/sdp Content-Length: 135 v=0 o=user1 53655765 2353687637 IN IP4 11.111.32.132 s=- c=IN IP4 11.111.32.132 t=0 0 m=audio 6000 RTP/AVP 0 a=rtpmap:0 PCMU/8000 <-------------> [Mar 2 16:08:45] DEBUG[30149] chan_sip.c: Header 0: INVITE sip:001555551636@app.fooobar.at:5160 SIP/2.0 (51) [Mar 2 16:08:45] DEBUG[30149] chan_sip.c: Header 1: Via: SIP/2.0/UDP 11.111.32.132:5061;branch=z9hG4bK-24391-1-4 (60) [Mar 2 16:08:45] DEBUG[30149] chan_sip.c: Header 2: From: ;tag=1 (50) [Mar 2 16:08:45] DEBUG[30149] chan_sip.c: Header 3: To: (42) [Mar 2 16:08:45] DEBUG[30149] chan_sip.c: Header 4: Proxy-Authorization: Digest username="u+431234567730151",realm="app.fooobar.at",uri="sip:22.22.222.184:5160",nonce="5ed78c32",response="438333757600bacc0aa1ca3f722821b4",algorithm=MD5 (183) [Mar 2 16:08:45] DEBUG[30149] chan_sip.c: Header 5: Call-ID: 1-24391@11.111.32.132 (30) [Mar 2 16:08:45] DEBUG[30149] chan_sip.c: Header 6: CSeq: 2 INVITE (14) [Mar 2 16:08:45] DEBUG[30149] chan_sip.c: Header 7: Contact: sip:sipp@11.111.32.132:5061 (36) [Mar 2 16:08:45] DEBUG[30149] chan_sip.c: Header 8: Max-Forwards: 70 (16) [Mar 2 16:08:45] DEBUG[30149] chan_sip.c: Header 9: Subject: Tests performed by darilion@ipcom.at (45) [Mar 2 16:08:45] DEBUG[30149] chan_sip.c: Header 10: Content-Type: application/sdp (29) [Mar 2 16:08:45] DEBUG[30149] chan_sip.c: Header 11: Content-Length: 135 (19) [Mar 2 16:08:45] DEBUG[30149] chan_sip.c: Header 12: (0) [Mar 2 16:08:45] DEBUG[30149] chan_sip.c: Line: v=0 (3) [Mar 2 16:08:45] DEBUG[30149] chan_sip.c: Line: o=user1 53655765 2353687637 IN IP4 11.111.32.132 (48) [Mar 2 16:08:45] DEBUG[30149] chan_sip.c: Line: s=- (3) [Mar 2 16:08:45] DEBUG[30149] chan_sip.c: Line: c=IN IP4 11.111.32.132 (22) [Mar 2 16:08:45] DEBUG[30149] chan_sip.c: Line: t=0 0 (5) [Mar 2 16:08:45] DEBUG[30149] chan_sip.c: Line: m=audio 6000 RTP/AVP 0 (22) [Mar 2 16:08:45] VERBOSE[30149] logger.c: --- (12 headers 7 lines) --- [Mar 2 16:08:45] DEBUG[30149] chan_sip.c: = Looking for Call ID: 1-24391@11.111.32.132 (Checking From) --From tag 1 --To-tag [Mar 2 16:08:45] DEBUG[30149] chan_sip.c: = Found Their Call ID: 1-24391@11.111.32.132 Their Tag 1 Our tag: as5f62ade5 [Mar 2 16:08:45] DEBUG[30149] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [Mar 2 16:08:45] VERBOSE[30149] logger.c: Sending to 11.111.32.132 : 5061 (NAT) [Mar 2 16:08:45] VERBOSE[30149] logger.c: Using INVITE request as basis request - 1-24391@11.111.32.132 [Mar 2 16:08:45] DEBUG[30149] chan_sip.c: Setting NAT on RTP to On [Mar 2 16:08:45] DEBUG[30149] chan_sip.c: Setting NAT on UDPTL to On [Mar 2 16:08:45] VERBOSE[30149] logger.c: Found user 'u+431234567730151' [Mar 2 16:08:45] VERBOSE[30149] logger.c: Found RTP audio format 0 [Mar 2 16:08:45] DEBUG[30149] chan_sip.c: Peer doesn't provide T.38 UDPTL [Mar 2 16:08:45] VERBOSE[30149] logger.c: Peer audio RTP is at port 11.111.32.132:6000 [Mar 2 16:08:45] VERBOSE[30149] logger.c: Found audio description format PCMU for ID 0 [Mar 2 16:08:45] DEBUG[30149] chan_sip.c: T38 state changed to 0 on channel [Mar 2 16:08:45] VERBOSE[30149] logger.c: Capabilities: us - 0x40e (gsm|ulaw|alaw|ilbc), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) [Mar 2 16:08:45] VERBOSE[30149] logger.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) [Mar 2 16:08:45] VERBOSE[30149] logger.c: Peer audio RTP is at port 11.111.32.132:6000 [Mar 2 16:08:45] DEBUG[30149] chan_sip.c: We're settling with these formats: 0x4 (ulaw) [Mar 2 16:08:45] DEBUG[30149] chan_sip.c: Checking SIP call limits for device u+431234567730151 [Mar 2 16:08:45] DEBUG[30149] chan_sip.c: Updating call counter for incoming call [Mar 2 16:08:45] DEBUG[30149] chan_sip.c: Call from peer 'u+431234567730151' is 1 out of 2 [Mar 2 16:08:45] DEBUG[30149] devicestate.c: Notification of state change to be queued on device/channel SIP/u+431234567730151 [Mar 2 16:08:45] DEBUG[30141] devicestate.c: No provider found, checking channel drivers for SIP - u+431234567730151 [Mar 2 16:08:45] VERBOSE[30149] logger.c: Looking for 001555551636 in fromSipPbx (domain app.fooobar.at) [Mar 2 16:08:45] DEBUG[30141] chan_sip.c: Checking device state for peer u+431234567730151 [Mar 2 16:08:45] DEBUG[30141] devicestate.c: Changing state for SIP/u+431234567730151 - state 2 (In use) [Mar 2 16:08:45] DEBUG[30149] chan_sip.c: *** Our native formats are 0x4 (ulaw) [Mar 2 16:08:45] DEBUG[30141] devicestate.c: No provider found, checking channel drivers for SIP - u+431234567730151 [Mar 2 16:08:45] DEBUG[30149] chan_sip.c: *** Joint capabilities are 0x4 (ulaw) [Mar 2 16:08:45] DEBUG[30141] chan_sip.c: Checking device state for peer u+431234567730151 [Mar 2 16:08:45] DEBUG[30164] app_queue.c: Device 'SIP/u+431234567730151' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Mar 2 16:08:45] DEBUG[30149] chan_sip.c: *** Our capabilities are 0x40e (gsm|ulaw|alaw|ilbc) [Mar 2 16:08:45] DEBUG[30149] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x4 (ulaw) [Mar 2 16:08:45] DEBUG[30149] chan_sip.c: This channel will not be able to handle video. [Mar 2 16:08:45] DEBUG[30149] chan_sip.c: build_route: Contact hop: sip:sipp@11.111.32.132:5061 [Mar 2 16:08:45] VERBOSE[30149] logger.c: list_route: hop: [Mar 2 16:08:45] DEBUG[30149] chan_sip.c: SIP/u+431234567730151-09e2e148: New call is still down.... Trying... [Mar 2 16:08:45] VERBOSE[30149] logger.c: <--- Transmitting (NAT) to 11.111.32.132:5061 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 11.111.32.132:5061;branch=z9hG4bK-24391-1-4;received=11.111.32.132 From: ;tag=1 To: Call-ID: 1-24391@11.111.32.132 CSeq: 2 INVITE User-Agent: MegaSIP-app Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Mar 2 16:08:45] DEBUG[30149] devicestate.c: Notification of state change to be queued on device/channel SIP/u+431234567730151 [Mar 2 16:08:45] DEBUG[30141] devicestate.c: No provider found, checking channel drivers for SIP - u+431234567730151 [Mar 2 16:08:45] DEBUG[30141] chan_sip.c: Checking device state for peer u+431234567730151 [Mar 2 16:08:45] DEBUG[15328] pbx.c: Launching 'DumpChan' [Mar 2 16:08:45] DEBUG[30141] devicestate.c: Changing state for SIP/u+431234567730151 - state 2 (In use) [Mar 2 16:08:45] VERBOSE[15328] logger.c: -- Executing [001555551636@fromSipPbx:1] DumpChan("SIP/u+431234567730151-09e2e148", "") in new stack [Mar 2 16:08:45] DEBUG[30164] app_queue.c: Device 'SIP/u+431234567730151' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Mar 2 16:08:45] DEBUG[30141] devicestate.c: No provider found, checking channel drivers for SIP - u+431234567730151 [Mar 2 16:08:45] DEBUG[30141] chan_sip.c: Checking device state for peer u+431234567730151 ... [dialplan commands] ... [Mar 2 16:08:45] DEBUG[15328] pbx.c: Launching 'Dial' [Mar 2 16:08:45] VERBOSE[15328] logger.c: -- Executing [s@macro-dialToGwAsteriskMakro:10] Dial("SIP/u+431234567730151-09e2e148", "SIP/+431555551636@gw-asterisk|300|i") in new stack [Mar 2 16:08:45] DEBUG[15328] chan_sip.c: Asked to create a SIP channel with formats: 0x4 (ulaw) [Mar 2 16:08:45] DEBUG[15328] chan_sip.c: Allocating new SIP dialog for (No Call-ID) - INVITE (With RTP) [Mar 2 16:08:45] DEBUG[15328] chan_sip.c: Our T38 capability (3856) [Mar 2 16:08:45] DEBUG[15328] chan_sip.c: Setting NAT on RTP to On [Mar 2 16:08:45] DEBUG[15328] chan_sip.c: Setting NAT on UDPTL to On [Mar 2 16:08:45] DEBUG[15328] chan_sip.c: *** Our native formats are 0x4 (ulaw) [Mar 2 16:08:45] DEBUG[15328] chan_sip.c: *** Joint capabilities are 0x0 (nothing) [Mar 2 16:08:45] DEBUG[15328] chan_sip.c: *** Our capabilities are 0x40e (gsm|ulaw|alaw|ilbc) [Mar 2 16:08:45] DEBUG[15328] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x4 (ulaw) [Mar 2 16:08:45] DEBUG[15328] chan_sip.c: *** Our preferred formats from the incoming channel are 0x4 (ulaw) [Mar 2 16:08:45] DEBUG[15328] chan_sip.c: This channel will not be able to handle video. [Mar 2 16:08:45] DEBUG[15328] channel.c: Not copying variable DIALEDTIME. [Mar 2 16:08:45] DEBUG[15328] channel.c: Not copying variable ANSWEREDTIME. [Mar 2 16:08:45] DEBUG[15328] channel.c: Not copying variable DIALEDPEERNAME. [Mar 2 16:08:45] DEBUG[15328] channel.c: Not copying variable DIALEDPEERNUMBER. [Mar 2 16:08:45] DEBUG[15328] channel.c: Not copying variable DIALSTATUS. [Mar 2 16:08:45] DEBUG[15328] channel.c: Not copying variable MACRO_DEPTH. [Mar 2 16:08:45] DEBUG[15328] channel.c: Not copying variable duration. [Mar 2 16:08:45] DEBUG[15328] channel.c: Not copying variable ext. [Mar 2 16:08:45] DEBUG[15328] channel.c: Not copying variable ARG2. [Mar 2 16:08:45] DEBUG[15328] channel.c: Not copying variable ARG1. [Mar 2 16:08:45] DEBUG[15328] channel.c: Not copying variable MACRO_PRIORITY. [Mar 2 16:08:45] DEBUG[15328] channel.c: Not copying variable MACRO_CONTEXT. [Mar 2 16:08:45] DEBUG[15328] channel.c: Not copying variable MACRO_EXTEN. [Mar 2 16:08:45] DEBUG[15328] channel.c: Copying hard-transferable variable SIPADDHEADER10. [Mar 2 16:08:45] DEBUG[15328] channel.c: Copying hard-transferable variable SIPADDHEADER09. [Mar 2 16:08:45] DEBUG[15328] channel.c: Copying hard-transferable variable SIPADDHEADER08. [Mar 2 16:08:45] DEBUG[15328] channel.c: Copying hard-transferable variable SIPADDHEADER07. [Mar 2 16:08:45] DEBUG[15328] channel.c: Copying hard-transferable variable SIPADDHEADER06. [Mar 2 16:08:45] DEBUG[15328] channel.c: Copying hard-transferable variable SIPADDHEADER05. [Mar 2 16:08:45] DEBUG[15328] channel.c: Copying hard-transferable variable SIPADDHEADER04. [Mar 2 16:08:45] DEBUG[15328] channel.c: Copying hard-transferable variable SIPADDHEADER03. [Mar 2 16:08:45] DEBUG[15328] channel.c: Copying hard-transferable variable SIPADDHEADER02. [Mar 2 16:08:45] DEBUG[15328] channel.c: Copying hard-transferable variable SIPADDHEADER01. [Mar 2 16:08:45] DEBUG[15328] channel.c: Not copying variable exten. [Mar 2 16:08:45] DEBUG[15328] channel.c: Not copying variable INVALID_EXTEN. [Mar 2 16:08:45] DEBUG[15328] channel.c: Not copying variable x-limit. [Mar 2 16:08:45] DEBUG[15328] channel.c: Not copying variable x-key. [Mar 2 16:08:45] DEBUG[15328] channel.c: Not copying variable SIPCALLID. [Mar 2 16:08:45] DEBUG[15328] channel.c: Not copying variable SIPDOMAIN. [Mar 2 16:08:45] DEBUG[15328] channel.c: Not copying variable SIPURI. [Mar 2 16:08:45] DEBUG[15328] chan_sip.c: Outgoing Call for +431555551636 [Mar 2 16:08:45] DEBUG[15328] chan_sip.c: Updating call counter for outgoing call [Mar 2 16:08:45] DEBUG[15328] chan_sip.c: Our T38 capability (3856), joint T38 capability (3856) [Mar 2 16:08:45] DEBUG[15328] chan_sip.c: ** Our capability: 0x40e (gsm|ulaw|alaw|ilbc) Video flag: False [Mar 2 16:08:45] DEBUG[15328] chan_sip.c: ** Our prefcodec: 0x4 (ulaw) [Mar 2 16:08:45] VERBOSE[15328] logger.c: Audio is at 22.22.222.184 port 10818 [Mar 2 16:08:45] VERBOSE[15328] logger.c: Adding codec 0x4 (ulaw) to SDP [Mar 2 16:08:45] VERBOSE[15328] logger.c: Adding codec 0x8 (alaw) to SDP [Mar 2 16:08:45] VERBOSE[15328] logger.c: Adding codec 0x400 (ilbc) to SDP [Mar 2 16:08:45] VERBOSE[15328] logger.c: Adding codec 0x2 (gsm) to SDP [Mar 2 16:08:45] VERBOSE[15328] logger.c: Adding non-codec 0x1 (telephone-event) to SDP [Mar 2 16:08:45] DEBUG[15328] chan_sip.c: -- Done with adding codecs to SDP [Mar 2 16:08:45] DEBUG[15328] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=-1) [Mar 2 16:08:45] DEBUG[15328] chan_sip.c: Done building SDP. Settling with this capability: 0x40e (gsm|ulaw|alaw|ilbc) [Mar 2 16:08:45] DEBUG[15328] chan_sip.c: Header 0: INVITE sip:+431555551636@22.22.222.183 SIP/2.0 (46) [Mar 2 16:08:45] DEBUG[15328] chan_sip.c: Header 1: Via: SIP/2.0/UDP 22.22.222.184:5160;branch=z9hG4bK622fc3f8;rport (64) [Mar 2 16:08:45] DEBUG[15328] chan_sip.c: Header 2: From: "+431234567730151" ;tag=as5e645e52 (80) [Mar 2 16:08:45] DEBUG[15328] chan_sip.c: Header 3: To: (37) [Mar 2 16:08:45] DEBUG[15328] chan_sip.c: Header 4: Contact: (49) [Mar 2 16:08:45] DEBUG[15328] chan_sip.c: Header 5: Call-ID: 0f31564b2befd78350413cfd4bbed686@22.22.222.184 (55) [Mar 2 16:08:45] DEBUG[15328] chan_sip.c: Header 6: CSeq: 102 INVITE (16) [Mar 2 16:08:45] DEBUG[15328] chan_sip.c: Header 7: User-Agent: MegaSIP-app (23) [Mar 2 16:08:45] DEBUG[15328] chan_sip.c: Header 8: Max-Forwards: 70 (16) [Mar 2 16:08:45] DEBUG[15328] chan_sip.c: Header 9: Date: Mon, 02 Mar 2009 15:08:45 GMT (35) [Mar 2 16:08:45] DEBUG[15328] chan_sip.c: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Mar 2 16:08:45] DEBUG[15328] chan_sip.c: Header 11: Supported: replaces (19) [Mar 2 16:08:45] DEBUG[15328] chan_sip.c: Header 22: Content-Type: application/sdp (29) [Mar 2 16:08:45] DEBUG[15328] chan_sip.c: Header 23: Content-Length: 334 (19) [Mar 2 16:08:45] DEBUG[15328] chan_sip.c: Header 24: (0) [Mar 2 16:08:45] DEBUG[15328] chan_sip.c: Line: v=0 (3) [Mar 2 16:08:45] DEBUG[15328] chan_sip.c: Line: o=root 30135 30135 IN IP4 22.22.222.184 (39) [Mar 2 16:08:45] DEBUG[15328] chan_sip.c: Line: s=session (9) [Mar 2 16:08:45] DEBUG[15328] chan_sip.c: Line: c=IN IP4 22.22.222.184 (22) [Mar 2 16:08:45] DEBUG[15328] chan_sip.c: Line: t=0 0 (5) [Mar 2 16:08:45] DEBUG[15328] chan_sip.c: Line: m=audio 10818 RTP/AVP 0 8 97 3 101 (34) [Mar 2 16:08:45] DEBUG[15328] chan_sip.c: Line: a=rtpmap:0 PCMU/8000 (20) [Mar 2 16:08:45] DEBUG[15328] chan_sip.c: Line: a=rtpmap:8 PCMA/8000 (20) [Mar 2 16:08:45] DEBUG[15328] chan_sip.c: Line: a=rtpmap:97 iLBC/8000 (21) [Mar 2 16:08:45] DEBUG[15328] chan_sip.c: Line: a=fmtp:97 mode=30 (17) [Mar 2 16:08:45] DEBUG[15328] chan_sip.c: Line: a=rtpmap:3 GSM/8000 (19) [Mar 2 16:08:45] DEBUG[15328] chan_sip.c: Line: a=rtpmap:101 telephone-event/8000 (33) [Mar 2 16:08:45] DEBUG[15328] chan_sip.c: Line: a=fmtp:101 0-16 (15) [Mar 2 16:08:45] DEBUG[15328] chan_sip.c: Line: a=silenceSupp:off - - - - (25) [Mar 2 16:08:45] DEBUG[15328] chan_sip.c: Line: a=ptime:20 (10) [Mar 2 16:08:45] DEBUG[15328] chan_sip.c: Line: a=sendrecv (10) [Mar 2 16:08:45] VERBOSE[15328] logger.c: Reliably Transmitting (NAT) to 22.22.222.183:5060: INVITE sip:+431555551636@22.22.222.183 SIP/2.0 Via: SIP/2.0/UDP 22.22.222.184:5160;branch=z9hG4bK622fc3f8;rport From: "+431234567730151" ;tag=as5e645e52 To: Contact: Call-ID: 0f31564b2befd78350413cfd4bbed686@22.22.222.184 CSeq: 102 INVITE User-Agent: MegaSIP-app Max-Forwards: 70 Date: Mon, 02 Mar 2009 15:08:45 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 334 v=0 o=root 30135 30135 IN IP4 22.22.222.184 s=session c=IN IP4 22.22.222.184 t=0 0 m=audio 10818 RTP/AVP 0 8 97 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Mar 2 16:08:45] DEBUG[15328] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #-1 [Mar 2 16:08:45] VERBOSE[15328] logger.c: -- Called +431555551636@gw-asterisk [Mar 2 16:08:45] VERBOSE[30149] logger.c: <--- SIP read from 22.22.222.183:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 22.22.222.184:5160;branch=z9hG4bK622fc3f8;received=22.22.222.184;rport=5160 From: "+431234567730151" ;tag=as5e645e52 To: Call-ID: 0f31564b2befd78350413cfd4bbed686@22.22.222.184 CSeq: 102 INVITE User-Agent: MegaSIP-gw Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <-------------> [Mar 2 16:08:45] DEBUG[30149] chan_sip.c: Header 0: SIP/2.0 100 Trying (18) [Mar 2 16:08:45] DEBUG[30149] chan_sip.c: Header 1: Via: SIP/2.0/UDP 22.22.222.184:5160;branch=z9hG4bK622fc3f8;received=22.22.222.184;rport=5160 (92) [Mar 2 16:08:45] DEBUG[30149] chan_sip.c: Header 2: From: "+431234567730151" ;tag=as5e645e52 (80) [Mar 2 16:08:45] DEBUG[30149] chan_sip.c: Header 3: To: (37) [Mar 2 16:08:45] DEBUG[30149] chan_sip.c: Header 4: Call-ID: 0f31564b2befd78350413cfd4bbed686@22.22.222.184 (55) [Mar 2 16:08:45] DEBUG[30149] chan_sip.c: Header 5: CSeq: 102 INVITE (16) [Mar 2 16:08:45] DEBUG[30149] chan_sip.c: Header 6: User-Agent: MegaSIP-gw (22) [Mar 2 16:08:45] DEBUG[30149] chan_sip.c: Header 7: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Mar 2 16:08:45] DEBUG[30149] chan_sip.c: Header 8: Supported: replaces (19) [Mar 2 16:08:45] DEBUG[30149] chan_sip.c: Header 9: Contact: (42) [Mar 2 16:08:45] DEBUG[30149] chan_sip.c: Header 10: Content-Length: 0 (17) [Mar 2 16:08:45] VERBOSE[30149] logger.c: --- (11 headers 0 lines) --- [Mar 2 16:08:45] DEBUG[30149] chan_sip.c: = Looking for Call ID: 0f31564b2befd78350413cfd4bbed686@22.22.222.184 (Checking To) --From tag as5e645e52 --To-tag [Mar 2 16:08:45] DEBUG[30149] chan_sip.c: = Found Their Call ID: 0f31564b2befd78350413cfd4bbed686@22.22.222.184 Their Tag Our tag: as5e645e52 [Mar 2 16:08:45] DEBUG[30149] chan_sip.c: *** SIP TIMER: Cancelling retransmission #81139 - INVITE (got response) [Mar 2 16:08:45] DEBUG[30149] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '0f31564b2befd78350413cfd4bbed686@22.22.222.184' Request 102: Found [Mar 2 16:08:45] DEBUG[30149] chan_sip.c: SIP response 100 to standard invite [Mar 2 16:08:47] VERBOSE[30149] logger.c: <--- SIP read from 22.22.222.183:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 22.22.222.184:5160;branch=z9hG4bK622fc3f8;received=22.22.222.184;rport=5160 From: "+431234567730151" ;tag=as5e645e52 To: ;tag=as712dde0c Call-ID: 0f31564b2befd78350413cfd4bbed686@22.22.222.184 CSeq: 102 INVITE User-Agent: MegaSIP-gw Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 289 v=0 o=root 18566 18566 IN IP4 22.22.222.183 s=session c=IN IP4 22.22.222.183 t=0 0 m=audio 11316 RTP/AVP 3 0 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <-------------> [Mar 2 16:08:47] DEBUG[30149] chan_sip.c: Header 0: SIP/2.0 183 Session Progress (28) [Mar 2 16:08:47] DEBUG[30149] chan_sip.c: Header 1: Via: SIP/2.0/UDP 22.22.222.184:5160;branch=z9hG4bK622fc3f8;received=22.22.222.184;rport=5160 (92) [Mar 2 16:08:47] DEBUG[30149] chan_sip.c: Header 2: From: "+431234567730151" ;tag=as5e645e52 (80) [Mar 2 16:08:47] DEBUG[30149] chan_sip.c: Header 3: To: ;tag=as712dde0c (52) [Mar 2 16:08:47] DEBUG[30149] chan_sip.c: Header 4: Call-ID: 0f31564b2befd78350413cfd4bbed686@22.22.222.184 (55) [Mar 2 16:08:47] DEBUG[30149] chan_sip.c: Header 5: CSeq: 102 INVITE (16) [Mar 2 16:08:47] DEBUG[30149] chan_sip.c: Header 6: User-Agent: MegaSIP-gw (22) [Mar 2 16:08:47] DEBUG[30149] chan_sip.c: Header 7: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Mar 2 16:08:47] DEBUG[30149] chan_sip.c: Header 8: Supported: replaces (19) [Mar 2 16:08:47] DEBUG[30149] chan_sip.c: Header 9: Contact: (42) [Mar 2 16:08:47] DEBUG[30149] chan_sip.c: Header 10: Content-Type: application/sdp (29) [Mar 2 16:08:47] DEBUG[30149] chan_sip.c: Header 11: Content-Length: 289 (19) [Mar 2 16:08:47] DEBUG[30149] chan_sip.c: Header 12: (0) [Mar 2 16:08:47] DEBUG[30149] chan_sip.c: Line: v=0 (3) [Mar 2 16:08:47] DEBUG[30149] chan_sip.c: Line: o=root 18566 18566 IN IP4 22.22.222.183 (39) [Mar 2 16:08:47] DEBUG[30149] chan_sip.c: Line: s=session (9) [Mar 2 16:08:47] DEBUG[30149] chan_sip.c: Line: c=IN IP4 22.22.222.183 (22) [Mar 2 16:08:47] DEBUG[30149] chan_sip.c: Line: t=0 0 (5) [Mar 2 16:08:47] DEBUG[30149] chan_sip.c: Line: m=audio 11316 RTP/AVP 3 0 8 101 (31) [Mar 2 16:08:47] DEBUG[30149] chan_sip.c: Line: a=rtpmap:3 GSM/8000 (19) [Mar 2 16:08:47] DEBUG[30149] chan_sip.c: Line: a=rtpmap:0 PCMU/8000 (20) [Mar 2 16:08:47] DEBUG[30149] chan_sip.c: Line: a=rtpmap:8 PCMA/8000 (20) [Mar 2 16:08:47] DEBUG[30149] chan_sip.c: Line: a=rtpmap:101 telephone-event/8000 (33) [Mar 2 16:08:47] DEBUG[30149] chan_sip.c: Line: a=fmtp:101 0-16 (15) [Mar 2 16:08:47] DEBUG[30149] chan_sip.c: Line: a=silenceSupp:off - - - - (25) [Mar 2 16:08:47] DEBUG[30149] chan_sip.c: Line: a=ptime:20 (10) [Mar 2 16:08:47] VERBOSE[30149] logger.c: --- (12 headers 14 lines) --- [Mar 2 16:08:47] DEBUG[30149] chan_sip.c: = Looking for Call ID: 0f31564b2befd78350413cfd4bbed686@22.22.222.184 (Checking To) --From tag as5e645e52 --To-tag as712dde0c [Mar 2 16:08:47] DEBUG[30149] chan_sip.c: = Found Their Call ID: 0f31564b2befd78350413cfd4bbed686@22.22.222.184 Their Tag Our tag: as5e645e52 [Mar 2 16:08:47] DEBUG[30149] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '0f31564b2befd78350413cfd4bbed686@22.22.222.184' Request 102: Found [Mar 2 16:08:47] DEBUG[30149] chan_sip.c: SIP response 183 to standard invite [Mar 2 16:08:47] VERBOSE[30149] logger.c: Found RTP audio format 3 [Mar 2 16:08:47] VERBOSE[30149] logger.c: Found RTP audio format 0 [Mar 2 16:08:47] VERBOSE[30149] logger.c: Found RTP audio format 8 [Mar 2 16:08:47] VERBOSE[30149] logger.c: Found RTP audio format 101 [Mar 2 16:08:47] DEBUG[30149] chan_sip.c: Peer doesn't provide T.38 UDPTL [Mar 2 16:08:47] VERBOSE[30149] logger.c: Peer audio RTP is at port 22.22.222.183:11316 [Mar 2 16:08:47] VERBOSE[30149] logger.c: Found audio description format GSM for ID 3 [Mar 2 16:08:47] VERBOSE[30149] logger.c: Found audio description format PCMU for ID 0 [Mar 2 16:08:47] VERBOSE[30149] logger.c: Found audio description format PCMA for ID 8 [Mar 2 16:08:47] VERBOSE[30149] logger.c: Found audio description format telephone-event for ID 101 [Mar 2 16:08:47] VERBOSE[30149] logger.c: Got unsupported a:fmtp in SDP offer [Mar 2 16:08:47] DEBUG[30149] chan_sip.c: T38 state changed to 0 on channel SIP/gw-asterisk-09e98798 [Mar 2 16:08:47] VERBOSE[30149] logger.c: Capabilities: us - 0x40e (gsm|ulaw|alaw|ilbc), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw) [Mar 2 16:08:47] VERBOSE[30149] logger.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Mar 2 16:08:47] VERBOSE[30149] logger.c: Peer audio RTP is at port 22.22.222.183:11316 [Mar 2 16:08:47] DEBUG[30149] chan_sip.c: We're settling with these formats: 0xe (gsm|ulaw|alaw) [Mar 2 16:08:47] DEBUG[30149] chan_sip.c: We have an owner, now see if we need to change this call [Mar 2 16:08:47] VERBOSE[15328] logger.c: -- SIP/gw-asterisk-09e98798 is making progress passing it to SIP/u+431234567730151-09e2e148 [Mar 2 16:08:47] DEBUG[15328] chan_sip.c: Setting framing from config on incoming call [Mar 2 16:08:47] DEBUG[15328] chan_sip.c: ** Our capability: 0x4 (ulaw) Video flag: True [Mar 2 16:08:47] DEBUG[15328] chan_sip.c: ** Our prefcodec: 0x0 (nothing) [Mar 2 16:08:47] VERBOSE[15328] logger.c: Audio is at 22.22.222.184 port 18282 [Mar 2 16:08:47] VERBOSE[15328] logger.c: Adding codec 0x4 (ulaw) to SDP [Mar 2 16:08:47] DEBUG[15328] chan_sip.c: -- Done with adding codecs to SDP [Mar 2 16:08:47] DEBUG[15328] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=-1) [Mar 2 16:08:47] DEBUG[15328] chan_sip.c: Done building SDP. Settling with this capability: 0x4 (ulaw) [Mar 2 16:08:47] VERBOSE[15328] logger.c: <--- Transmitting (NAT) to 11.111.32.132:5061 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 11.111.32.132:5061;branch=z9hG4bK-24391-1-4;received=11.111.32.132 From: ;tag=1 To: ;tag=as5bcc327b Call-ID: 1-24391@11.111.32.132 CSeq: 2 INVITE User-Agent: MegaSIP-app Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 186 v=0 o=root 30135 30135 IN IP4 22.22.222.184 s=session c=IN IP4 22.22.222.184 t=0 0 m=audio 18282 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> [Mar 2 16:08:47] DEBUG[15328] rtp.c: Ooh, format changed from unknown to ulaw [Mar 2 16:08:47] DEBUG[15328] rtp.c: Created smoother: format: 4 ms: 20 len: 160 [Mar 2 16:08:47] VERBOSE[30149] logger.c: <--- SIP read from 22.22.222.183:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 22.22.222.184:5160;branch=z9hG4bK622fc3f8;received=22.22.222.184;rport=5160 From: "+431234567730151" ;tag=as5e645e52 To: ;tag=as712dde0c Call-ID: 0f31564b2befd78350413cfd4bbed686@22.22.222.184 CSeq: 102 INVITE User-Agent: MegaSIP-gw Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <-------------> [Mar 2 16:08:47] DEBUG[30149] chan_sip.c: Header 0: SIP/2.0 180 Ringing (19) [Mar 2 16:08:47] DEBUG[30149] chan_sip.c: Header 1: Via: SIP/2.0/UDP 22.22.222.184:5160;branch=z9hG4bK622fc3f8;received=22.22.222.184;rport=5160 (92) [Mar 2 16:08:47] DEBUG[30149] chan_sip.c: Header 2: From: "+431234567730151" ;tag=as5e645e52 (80) [Mar 2 16:08:47] DEBUG[30149] chan_sip.c: Header 3: To: ;tag=as712dde0c (52) [Mar 2 16:08:47] DEBUG[30149] chan_sip.c: Header 4: Call-ID: 0f31564b2befd78350413cfd4bbed686@22.22.222.184 (55) [Mar 2 16:08:47] DEBUG[30149] chan_sip.c: Header 5: CSeq: 102 INVITE (16) [Mar 2 16:08:47] DEBUG[30149] chan_sip.c: Header 6: User-Agent: MegaSIP-gw (22) [Mar 2 16:08:47] DEBUG[30149] chan_sip.c: Header 7: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Mar 2 16:08:47] DEBUG[30149] chan_sip.c: Header 8: Supported: replaces (19) [Mar 2 16:08:47] DEBUG[30149] chan_sip.c: Header 9: Contact: (42) [Mar 2 16:08:47] DEBUG[30149] chan_sip.c: Header 10: Content-Length: 0 (17) [Mar 2 16:08:47] VERBOSE[30149] logger.c: --- (11 headers 0 lines) --- [Mar 2 16:08:47] DEBUG[30149] chan_sip.c: = Looking for Call ID: 0f31564b2befd78350413cfd4bbed686@22.22.222.184 (Checking To) --From tag as5e645e52 --To-tag as712dde0c [Mar 2 16:08:47] DEBUG[30149] chan_sip.c: = Found Their Call ID: 0f31564b2befd78350413cfd4bbed686@22.22.222.184 Their Tag as712dde0c Our tag: as5e645e52 [Mar 2 16:08:47] DEBUG[30149] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '0f31564b2befd78350413cfd4bbed686@22.22.222.184' Request 102: Found [Mar 2 16:08:47] DEBUG[30149] chan_sip.c: SIP response 180 to standard invite [Mar 2 16:08:47] DEBUG[30149] devicestate.c: Notification of state change to be queued on device/channel SIP/gw-asterisk [Mar 2 16:08:47] DEBUG[30141] devicestate.c: No provider found, checking channel drivers for SIP - gw-asterisk [Mar 2 16:08:47] DEBUG[30141] chan_sip.c: Checking device state for peer gw-asterisk [Mar 2 16:08:47] DEBUG[30141] devicestate.c: Changing state for SIP/gw-asterisk - state 1 (Not in use) [Mar 2 16:08:47] DEBUG[30164] app_queue.c: Device 'SIP/gw-asterisk' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Mar 2 16:08:47] VERBOSE[15328] logger.c: -- SIP/gw-asterisk-09e98798 is ringing [Mar 2 16:08:47] VERBOSE[15328] logger.c: <--- Transmitting (NAT) to 11.111.32.132:5061 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 11.111.32.132:5061;branch=z9hG4bK-24391-1-4;received=11.111.32.132 From: ;tag=1 To: ;tag=as5bcc327b Call-ID: 1-24391@11.111.32.132 CSeq: 2 INVITE User-Agent: MegaSIP-app Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Mar 2 16:08:47] VERBOSE[30149] logger.c: <--- SIP read from 11.111.32.132:5061 ---> CANCEL sip:001555551636@app.fooobar.at:5160 SIP/2.0 Via: SIP/2.0/UDP 11.111.32.132:5061;branch=z9hG4bK-24391-1-4 From: ;tag=1 To: Call-ID: 1-24391@11.111.32.132 CSeq: 2 CANCEL Max-Forwards: 70 <-------------> [Mar 2 16:08:47] DEBUG[30149] chan_sip.c: Header 0: CANCEL sip:001555551636@app.fooobar.at:5160 SIP/2.0 (51) [Mar 2 16:08:47] DEBUG[30149] chan_sip.c: Header 1: Via: SIP/2.0/UDP 11.111.32.132:5061;branch=z9hG4bK-24391-1-4 (60) [Mar 2 16:08:47] DEBUG[30149] chan_sip.c: Header 2: From: ;tag=1 (50) [Mar 2 16:08:47] DEBUG[30149] chan_sip.c: Header 3: To: (42) [Mar 2 16:08:47] DEBUG[30149] chan_sip.c: Header 4: Call-ID: 1-24391@11.111.32.132 (30) [Mar 2 16:08:47] DEBUG[30149] chan_sip.c: Header 5: CSeq: 2 CANCEL (14) [Mar 2 16:08:47] DEBUG[30149] chan_sip.c: Header 6: Max-Forwards: 70 (16) [Mar 2 16:08:47] VERBOSE[30149] logger.c: --- (7 headers 0 lines) --- [Mar 2 16:08:47] DEBUG[30149] chan_sip.c: = Looking for Call ID: 1-24391@11.111.32.132 (Checking From) --From tag 1 --To-tag [Mar 2 16:08:47] DEBUG[30149] chan_sip.c: = No match Their Call ID: 0f31564b2befd78350413cfd4bbed686@22.22.222.184 Their Tag as712dde0c Our tag: as5e645e52 [Mar 2 16:08:47] DEBUG[30149] chan_sip.c: = Found Their Call ID: 1-24391@11.111.32.132 Their Tag 1 Our tag: as5bcc327b [Mar 2 16:08:47] DEBUG[30149] chan_sip.c: **** Received CANCEL (14) - Command in SIP CANCEL [Mar 2 16:08:47] VERBOSE[30149] logger.c: Sending to 11.111.32.132 : 5061 (NAT) [Mar 2 16:08:47] DEBUG[30149] chan_sip.c: Setting SIP_ALREADYGONE on dialog 1-24391@11.111.32.132 [Mar 2 16:08:47] DEBUG[30149] chan_sip.c: Updating call counter for incoming call [Mar 2 16:08:47] DEBUG[30149] chan_sip.c: Call from peer 'u+431234567730151' removed from call limit 2 [Mar 2 16:08:47] DEBUG[30149] devicestate.c: Notification of state change to be queued on device/channel SIP/u+431234567730151 [Mar 2 16:08:47] DEBUG[30141] devicestate.c: No provider found, checking channel drivers for SIP - u+431234567730151 [Mar 2 16:08:47] DEBUG[30141] chan_sip.c: Checking device state for peer u+431234567730151 [Mar 2 16:08:47] VERBOSE[30149] logger.c: <--- Reliably Transmitting (NAT) to 11.111.32.132:5061 ---> SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 11.111.32.132:5061;branch=z9hG4bK-24391-1-4;received=11.111.32.132 From: ;tag=1 To: ;tag=as5bcc327b Call-ID: 1-24391@11.111.32.132 CSeq: 2 INVITE User-Agent: MegaSIP-app Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 <------------> [Mar 2 16:08:47] DEBUG[30141] devicestate.c: Changing state for SIP/u+431234567730151 - state 1 (Not in use) [Mar 2 16:08:47] DEBUG[30149] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #-1 [Mar 2 16:08:47] DEBUG[30141] devicestate.c: No provider found, checking channel drivers for SIP - u+431234567730151 [Mar 2 16:08:47] DEBUG[30141] chan_sip.c: Checking device state for peer u+431234567730151 [Mar 2 16:08:47] DEBUG[30164] app_queue.c: Device 'SIP/u+431234567730151' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Mar 2 16:08:47] VERBOSE[30149] logger.c: <--- Transmitting (NAT) to 11.111.32.132:5061 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 11.111.32.132:5061;branch=z9hG4bK-24391-1-4;received=11.111.32.132 From: ;tag=1 To: ;tag=as5bcc327b Call-ID: 1-24391@11.111.32.132 CSeq: 2 CANCEL User-Agent: MegaSIP-app Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Mar 2 16:08:47] DEBUG[15328] rtp.c: Channel '' has no RTP, not doing anything [Mar 2 16:08:47] DEBUG[15328] channel.c: Hanging up channel 'SIP/gw-asterisk-09e98798' [Mar 2 16:08:47] DEBUG[15328] chan_sip.c: Hangup call SIP/gw-asterisk-09e98798, SIP callid 0f31564b2befd78350413cfd4bbed686@22.22.222.184) [Mar 2 16:08:47] DEBUG[15328] chan_sip.c: Hanging up channel in state Ringing (not UP) [Mar 2 16:08:47] VERBOSE[15328] logger.c: Scheduling destruction of SIP dialog '0f31564b2befd78350413cfd4bbed686@22.22.222.184' in 32000 ms (Method: INVITE) [Mar 2 16:08:47] DEBUG[15328] chan_sip.c: Acked pending invite 102 [Mar 2 16:08:47] DEBUG[15328] chan_sip.c: Stopping retransmission on '0f31564b2befd78350413cfd4bbed686@22.22.222.184' of Request 102: Match Found [Mar 2 16:08:47] VERBOSE[15328] logger.c: Reliably Transmitting (NAT) to 22.22.222.183:5060: CANCEL sip:+431555551636@22.22.222.183 SIP/2.0 Via: SIP/2.0/UDP 22.22.222.184:5160;branch=z9hG4bK622fc3f8;rport From: "+431234567730151" ;tag=as5e645e52 To: Call-ID: 0f31564b2befd78350413cfd4bbed686@22.22.222.184 CSeq: 102 CANCEL User-Agent: MegaSIP-app Max-Forwards: 70 Content-Length: 0 --- [Mar 2 16:08:47] DEBUG[15328] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #-1 [Mar 2 16:08:47] VERBOSE[15328] logger.c: Scheduling destruction of SIP dialog '0f31564b2befd78350413cfd4bbed686@22.22.222.184' in 32000 ms (Method: INVITE) [Mar 2 16:08:47] DEBUG[15328] devicestate.c: Notification of state change to be queued on device/channel SIP/gw-asterisk [Mar 2 16:08:47] DEBUG[15328] app_dial.c: Exiting with DIALSTATUS=CANCEL. [Mar 2 16:08:47] DEBUG[30141] devicestate.c: No provider found, checking channel drivers for SIP - gw-asterisk [Mar 2 16:08:47] DEBUG[30141] chan_sip.c: Checking device state for peer gw-asterisk [Mar 2 16:08:47] DEBUG[15328] app_macro.c: Spawn extension (macro-dialToGwAsteriskMakro,s,10) exited non-zero on 'SIP/u+431234567730151-09e2e148' in macro 'dialToGwAsteriskMakro' [Mar 2 16:08:47] DEBUG[30141] devicestate.c: Changing state for SIP/gw-asterisk - state 1 (Not in use) [Mar 2 16:08:47] DEBUG[15328] pbx.c: Spawn extension (fromSipToPstn,+431555551636,7) exited non-zero on 'SIP/u+431234567730151-09e2e148' [Mar 2 16:08:47] DEBUG[30164] app_queue.c: Device 'SIP/gw-asterisk' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Mar 2 16:08:47] VERBOSE[15328] logger.c: == Spawn extension (fromSipToPstn, +431555551636, 7) exited non-zero on 'SIP/u+431234567730151-09e2e148' [Mar 2 16:08:47] VERBOSE[30149] logger.c: <--- SIP read from 22.22.222.183:5060 ---> SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 22.22.222.184:5160;branch=z9hG4bK622fc3f8;received=22.22.222.184;rport=5160 From: "+431234567730151" ;tag=as5e645e52 To: ;tag=as712dde0c Call-ID: 0f31564b2befd78350413cfd4bbed686@22.22.222.184 CSeq: 102 INVITE User-Agent: MegaSIP-gw Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 <-------------> [Mar 2 16:08:47] DEBUG[15328] channel.c: Soft-Hanging up channel 'SIP/u+431234567730151-09e2e148' [Mar 2 16:08:47] DEBUG[30149] chan_sip.c: Header 0: SIP/2.0 487 Request Terminated (30) [Mar 2 16:08:47] DEBUG[15328] channel.c: Hanging up channel 'SIP/u+431234567730151-09e2e148' [Mar 2 16:08:47] DEBUG[15328] chan_sip.c: Hangup call SIP/u+431234567730151-09e2e148, SIP callid 1-24391@11.111.32.132) [Mar 2 16:08:47] DEBUG[30149] chan_sip.c: Header 1: Via: SIP/2.0/UDP 22.22.222.184:5160;branch=z9hG4bK622fc3f8;received=22.22.222.184;rport=5160 (92) [Mar 2 16:08:47] DEBUG[15328] chan_sip.c: Hanging up channel in state Ring (not UP) [Mar 2 16:08:47] DEBUG[30149] chan_sip.c: Header 2: From: "+431234567730151" ;tag=as5e645e52 (80) [Mar 2 16:08:47] DEBUG[30149] chan_sip.c: Header 3: To: ;tag=as712dde0c (52) [Mar 2 16:08:47] DEBUG[30149] chan_sip.c: Header 4: Call-ID: 0f31564b2befd78350413cfd4bbed686@22.22.222.184 (55) [Mar 2 16:08:47] DEBUG[30149] chan_sip.c: Header 5: CSeq: 102 INVITE (16) [Mar 2 16:08:47] DEBUG[30149] chan_sip.c: Header 6: User-Agent: MegaSIP-gw (22) [Mar 2 16:08:47] DEBUG[30149] chan_sip.c: Header 7: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Mar 2 16:08:47] DEBUG[30149] chan_sip.c: Header 8: Supported: replaces (19) [Mar 2 16:08:47] DEBUG[15328] devicestate.c: Notification of state change to be queued on device/channel SIP/u+431234567730151 [Mar 2 16:08:47] DEBUG[30149] chan_sip.c: Header 9: Content-Length: 0 (17) [Mar 2 16:08:47] DEBUG[30141] devicestate.c: No provider found, checking channel drivers for SIP - u+431234567730151 [Mar 2 16:08:47] VERBOSE[30149] logger.c: --- (10 headers 0 lines) --- [Mar 2 16:08:47] DEBUG[30141] chan_sip.c: Checking device state for peer u+431234567730151 [Mar 2 16:08:47] DEBUG[30149] chan_sip.c: = Looking for Call ID: 0f31564b2befd78350413cfd4bbed686@22.22.222.184 (Checking To) --From tag as5e645e52 --To-tag as712dde0c [Mar 2 16:08:47] DEBUG[30141] devicestate.c: Changing state for SIP/u+431234567730151 - state 1 (Not in use) [Mar 2 16:08:47] DEBUG[30149] chan_sip.c: = Found Their Call ID: 0f31564b2befd78350413cfd4bbed686@22.22.222.184 Their Tag as712dde0c Our tag: as5e645e52 [Mar 2 16:08:47] DEBUG[30141] devicestate.c: No provider found, checking channel drivers for SIP - u+431234567730151 [Mar 2 16:08:47] DEBUG[30149] chan_sip.c: Stopping retransmission on '0f31564b2befd78350413cfd4bbed686@22.22.222.184' of Request 102: Match Not Found [Mar 2 16:08:47] DEBUG[30141] chan_sip.c: Checking device state for peer u+431234567730151 [Mar 2 16:08:47] DEBUG[30164] app_queue.c: Device 'SIP/u+431234567730151' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Mar 2 16:08:47] DEBUG[30149] chan_sip.c: SIP response 487 to standard invite [Mar 2 16:08:47] VERBOSE[30149] logger.c: Transmitting (NAT) to 22.22.222.183:5060: ACK sip:+431555551636@22.22.222.183 SIP/2.0 Via: SIP/2.0/UDP 22.22.222.184:5160;branch=z9hG4bK622fc3f8;rport From: "+431234567730151" ;tag=as5e645e52 To: ;tag=as712dde0c Contact: Call-ID: 0f31564b2befd78350413cfd4bbed686@22.22.222.184 CSeq: 102 ACK User-Agent: MegaSIP-app Max-Forwards: 70 Content-Length: 0 --- [Mar 2 16:08:47] DEBUG[30149] chan_sip.c: Updating call counter for outgoing call [Mar 2 16:08:47] DEBUG[30149] chan_sip.c: Setting SIP_ALREADYGONE on dialog 0f31564b2befd78350413cfd4bbed686@22.22.222.184 [Mar 2 16:08:47] VERBOSE[30149] logger.c: <--- SIP read from 22.22.222.183:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 22.22.222.184:5160;branch=z9hG4bK622fc3f8;received=22.22.222.184;rport=5160 From: "+431234567730151" ;tag=as5e645e52 To: ;tag=as712dde0c Call-ID: 0f31564b2befd78350413cfd4bbed686@22.22.222.184 CSeq: 102 CANCEL User-Agent: MegaSIP-gw Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <-------------> [Mar 2 16:08:47] DEBUG[30149] chan_sip.c: Header 0: SIP/2.0 200 OK (14) [Mar 2 16:08:47] DEBUG[30149] chan_sip.c: Header 1: Via: SIP/2.0/UDP 22.22.222.184:5160;branch=z9hG4bK622fc3f8;received=22.22.222.184;rport=5160 (92) [Mar 2 16:08:47] DEBUG[30149] chan_sip.c: Header 2: From: "+431234567730151" ;tag=as5e645e52 (80) [Mar 2 16:08:47] DEBUG[30149] chan_sip.c: Header 3: To: ;tag=as712dde0c (52) [Mar 2 16:08:47] DEBUG[30149] chan_sip.c: Header 4: Call-ID: 0f31564b2befd78350413cfd4bbed686@22.22.222.184 (55) [Mar 2 16:08:47] DEBUG[30149] chan_sip.c: Header 5: CSeq: 102 CANCEL (16) [Mar 2 16:08:47] DEBUG[30149] chan_sip.c: Header 6: User-Agent: MegaSIP-gw (22) [Mar 2 16:08:47] DEBUG[30149] chan_sip.c: Header 7: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Mar 2 16:08:47] DEBUG[30149] chan_sip.c: Header 8: Supported: replaces (19) [Mar 2 16:08:47] DEBUG[30149] chan_sip.c: Header 9: Contact: (42) [Mar 2 16:08:47] DEBUG[30149] chan_sip.c: Header 10: Content-Length: 0 (17) [Mar 2 16:08:47] VERBOSE[30149] logger.c: --- (11 headers 0 lines) --- [Mar 2 16:08:47] DEBUG[30149] chan_sip.c: = Looking for Call ID: 0f31564b2befd78350413cfd4bbed686@22.22.222.184 (Checking To) --From tag as5e645e52 --To-tag as712dde0c [Mar 2 16:08:47] DEBUG[30149] chan_sip.c: = Found Their Call ID: 0f31564b2befd78350413cfd4bbed686@22.22.222.184 Their Tag as712dde0c Our tag: as5e645e52 [Mar 2 16:08:47] DEBUG[30149] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #81145 [Mar 2 16:08:47] DEBUG[30149] chan_sip.c: Stopping retransmission on '0f31564b2befd78350413cfd4bbed686@22.22.222.184' of Request 102: Match Found [Mar 2 16:08:47] VERBOSE[30149] logger.c: Really destroying SIP dialog '0f31564b2befd78350413cfd4bbed686@22.22.222.184' Method: INVITE [Mar 2 16:08:47] VERBOSE[30149] logger.c: <--- SIP read from 11.111.32.132:5061 ---> CANCEL sip:001555551636@app.fooobar.at:5160 SIP/2.0 Via: SIP/2.0/UDP 11.111.32.132:5061;branch=z9hG4bK-24391-1-5 From: ;tag=1 To: Call-ID: 1-24391@11.111.32.132 CSeq: 2 CANCEL Max-Forwards: 70 <-------------> [Mar 2 16:08:47] DEBUG[30149] chan_sip.c: Header 0: CANCEL sip:001555551636@app.fooobar.at:5160 SIP/2.0 (51) [Mar 2 16:08:47] DEBUG[30149] chan_sip.c: Header 1: Via: SIP/2.0/UDP 11.111.32.132:5061;branch=z9hG4bK-24391-1-5 (60) [Mar 2 16:08:47] DEBUG[30149] chan_sip.c: Header 2: From: ;tag=1 (50) [Mar 2 16:08:47] DEBUG[30149] chan_sip.c: Header 3: To: (42) [Mar 2 16:08:47] DEBUG[30149] chan_sip.c: Header 4: Call-ID: 1-24391@11.111.32.132 (30) [Mar 2 16:08:47] DEBUG[30149] chan_sip.c: Header 5: CSeq: 2 CANCEL (14) [Mar 2 16:08:47] DEBUG[30149] chan_sip.c: Header 6: Max-Forwards: 70 (16) [Mar 2 16:08:47] VERBOSE[30149] logger.c: --- (7 headers 0 lines) --- [Mar 2 16:08:47] DEBUG[30149] chan_sip.c: = Looking for Call ID: 1-24391@11.111.32.132 (Checking From) --From tag 1 --To-tag [Mar 2 16:08:47] DEBUG[30149] chan_sip.c: = Found Their Call ID: 1-24391@11.111.32.132 Their Tag 1 Our tag: as5bcc327b [Mar 2 16:08:47] DEBUG[30149] chan_sip.c: **** Received CANCEL (14) - Command in SIP CANCEL [Mar 2 16:08:47] DEBUG[30149] chan_sip.c: Ignoring SIP message because of retransmit (CANCEL Seqno 2, ours 2) [Mar 2 16:08:47] VERBOSE[30149] logger.c: Sending to 11.111.32.132 : 5061 (NAT) [Mar 2 16:08:47] DEBUG[30149] chan_sip.c: Setting SIP_ALREADYGONE on dialog 1-24391@11.111.32.132 [Mar 2 16:08:47] VERBOSE[30149] logger.c: Scheduling destruction of SIP dialog '1-24391@11.111.32.132' in 32000 ms (Method: CANCEL) [Mar 2 16:08:47] VERBOSE[30149] logger.c: <--- Reliably Transmitting (NAT) to 11.111.32.132:5061 ---> SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 11.111.32.132:5061;branch=z9hG4bK-24391-1-4;received=11.111.32.132 From: ;tag=1 To: ;tag=as5bcc327b Call-ID: 1-24391@11.111.32.132 CSeq: 2 INVITE User-Agent: MegaSIP-app Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 <------------> [Mar 2 16:08:47] DEBUG[30149] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #-1 [Mar 2 16:08:47] VERBOSE[30149] logger.c: <--- Transmitting (NAT) to 11.111.32.132:5061 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 11.111.32.132:5061;branch=z9hG4bK-24391-1-5;received=11.111.32.132 From: ;tag=1 To: ;tag=as5bcc327b Call-ID: 1-24391@11.111.32.132 CSeq: 2 CANCEL User-Agent: MegaSIP-app Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Mar 2 16:08:47] VERBOSE[30149] logger.c: <--- SIP read from 11.111.32.132:5061 ---> ACK sip:001555551636@app.fooobar.at:5160 SIP/2.0 Via: SIP/2.0/UDP 11.111.32.132:5061;branch=z9hG4bK-24391-1-5 From: ;tag=1 To: ;tag=as5bcc327b Call-ID: 1-24391@11.111.32.132 CSeq: 2 ACK Max-Forwards: 70 <-------------> [Mar 2 16:08:47] DEBUG[30149] chan_sip.c: Header 0: ACK sip:001555551636@app.fooobar.at:5160 SIP/2.0 (48) [Mar 2 16:08:47] DEBUG[30149] chan_sip.c: Header 1: Via: SIP/2.0/UDP 11.111.32.132:5061;branch=z9hG4bK-24391-1-5 (60) [Mar 2 16:08:47] DEBUG[30149] chan_sip.c: Header 2: From: ;tag=1 (50) [Mar 2 16:08:47] DEBUG[30149] chan_sip.c: Header 3: To: ;tag=as5bcc327b (57) [Mar 2 16:08:47] DEBUG[30149] chan_sip.c: Header 4: Call-ID: 1-24391@11.111.32.132 (30) [Mar 2 16:08:47] DEBUG[30149] chan_sip.c: Header 5: CSeq: 2 ACK (11) [Mar 2 16:08:47] DEBUG[30149] chan_sip.c: Header 6: Max-Forwards: 70 (16) [Mar 2 16:08:47] VERBOSE[30149] logger.c: --- (7 headers 0 lines) --- [Mar 2 16:08:47] DEBUG[30149] chan_sip.c: = Looking for Call ID: 1-24391@11.111.32.132 (Checking From) --From tag 1 --To-tag as5bcc327b [Mar 2 16:08:47] DEBUG[30149] chan_sip.c: = Found Their Call ID: 1-24391@11.111.32.132 Their Tag 1 Our tag: as5bcc327b [Mar 2 16:08:47] DEBUG[30149] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [Mar 2 16:08:47] DEBUG[30149] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #81148 [Mar 2 16:08:47] DEBUG[30149] chan_sip.c: Stopping retransmission on '1-24391@11.111.32.132' of Response 2: Match Found [Mar 2 16:08:48] DEBUG[30149] chan_sip.c: SIP TIMER: Rescheduling retransmission #81143 (1) SIP/2.0 - 1 [Mar 2 16:08:48] DEBUG[30149] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 1000 ms (t1 500 ms (Retrans id #81143)) [Mar 2 16:08:48] VERBOSE[30149] logger.c: Retransmitting #1 (NAT) to 11.111.32.132:5061: SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 11.111.32.132:5061;branch=z9hG4bK-24391-1-4;received=11.111.32.132 From: ;tag=1 To: ;tag=as5bcc327b Call-ID: 1-24391@11.111.32.132 CSeq: 2 INVITE User-Agent: MegaSIP-app Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Mar 2 16:08:49] DEBUG[30149] chan_sip.c: SIP TIMER: Rescheduling retransmission #81143 (2) SIP/2.0 - 1 [Mar 2 16:08:49] DEBUG[30149] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 2000 ms (t1 500 ms (Retrans id #81143)) [Mar 2 16:08:49] VERBOSE[30149] logger.c: Retransmitting #2 (NAT) to 11.111.32.132:5061: SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 11.111.32.132:5061;branch=z9hG4bK-24391-1-4;received=11.111.32.132 From: ;tag=1 To: ;tag=as5bcc327b Call-ID: 1-24391@11.111.32.132 CSeq: 2 INVITE User-Agent: MegaSIP-app Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Mar 2 16:08:50] VERBOSE[30149] logger.c: <--- SIP read from 11.111.33.3:39480 ---> <-------------> [Mar 2 16:08:50] DEBUG[30149] chan_sip.c: Header 0: (0) [Mar 2 16:08:51] DEBUG[30149] chan_sip.c: SIP TIMER: Rescheduling retransmission #81143 (3) SIP/2.0 - 1 [Mar 2 16:08:51] DEBUG[30149] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 4000 ms (t1 500 ms (Retrans id #81143)) [Mar 2 16:08:51] VERBOSE[30149] logger.c: Retransmitting #3 (NAT) to 11.111.32.132:5061: SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 11.111.32.132:5061;branch=z9hG4bK-24391-1-4;received=11.111.32.132 From: ;tag=1 To: ;tag=as5bcc327b Call-ID: 1-24391@11.111.32.132 CSeq: 2 INVITE User-Agent: MegaSIP-app Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Mar 2 16:08:55] DEBUG[30149] chan_sip.c: SIP TIMER: Rescheduling retransmission #81143 (4) SIP/2.0 - 1 [Mar 2 16:08:55] DEBUG[30149] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 4000 ms (t1 500 ms (Retrans id #81143)) [Mar 2 16:08:55] VERBOSE[30149] logger.c: Retransmitting #4 (NAT) to 11.111.32.132:5061: SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 11.111.32.132:5061;branch=z9hG4bK-24391-1-4;received=11.111.32.132 From: ;tag=1 To: ;tag=as5bcc327b Call-ID: 1-24391@11.111.32.132 CSeq: 2 INVITE User-Agent: MegaSIP-app Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Mar 2 16:08:59] DEBUG[30149] chan_sip.c: SIP TIMER: Rescheduling retransmission #81143 (5) SIP/2.0 - 1 [Mar 2 16:08:59] DEBUG[30149] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 4000 ms (t1 500 ms (Retrans id #81143)) [Mar 2 16:08:59] VERBOSE[30149] logger.c: Retransmitting #5 (NAT) to 11.111.32.132:5061: SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 11.111.32.132:5061;branch=z9hG4bK-24391-1-4;received=11.111.32.132 From: ;tag=1 To: ;tag=as5bcc327b Call-ID: 1-24391@11.111.32.132 CSeq: 2 INVITE User-Agent: MegaSIP-app Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Mar 2 16:09:03] DEBUG[30149] chan_sip.c: SIP TIMER: Rescheduling retransmission #81143 (6) SIP/2.0 - 1 [Mar 2 16:09:03] DEBUG[30149] chan_sip.c: ** SIP timers: Rescheduling retransmission 7 to 4000 ms (t1 500 ms (Retrans id #81143)) [Mar 2 16:09:03] VERBOSE[30149] logger.c: Retransmitting #6 (NAT) to 11.111.32.132:5061: SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 11.111.32.132:5061;branch=z9hG4bK-24391-1-4;received=11.111.32.132 From: ;tag=1 To: ;tag=as5bcc327b Call-ID: 1-24391@11.111.32.132 CSeq: 2 INVITE User-Agent: MegaSIP-app Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Mar 2 16:09:07] WARNING[30149] chan_sip.c: Maximum retries exceeded on transmission 1-24391@11.111.32.132 for seqno 2 (Critical Response) -- See doc/sip-retransmit.txt. [Mar 2 16:09:07] DEBUG[30149] chan_sip.c: Setting SIP_ALREADYGONE on dialog 1-24391@11.111.32.132 [Mar 2 16:09:07] VERBOSE[30149] logger.c: Really destroying SIP dialog '1-24391@11.111.32.132' Method: ACK ^C