| Summary: | ASTERISK-18493: One size sound | ||||
| Reporter: | Badalian Vyacheslav (slavon) | Labels: | |||
| Date Opened: | 2011-09-09 08:35:08 | Date Closed: | 2011-09-12 12:52:37 | ||
| Priority: | Major | Regression? | |||
| Status: | Closed/Complete | Components: | Channels/chan_sip/General | ||
| Versions: | 1.8.6.0 | Frequency of Occurrence | |||
| Related Issues: |
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| Environment: | Attachments: | ||||
| Description: | == Spawn extension (office, 200, 1) exited non-zero on 'SIP/111_office-000000e3' [Sep 9 17:29:33] WARNING[21474]: chan_sip.c:3349 __sip_xmit: sip_xmit of 0x1bb8fbc0 (len 574) to 0.0.0.111:5060 returned -1: Invalid argument [Sep 9 17:29:33] WARNING[11319]: chan_sip.c:3349 __sip_xmit: sip_xmit of 0x1bb8fbc0 (len 574) to 0.0.0.111:5060 returned -1: Invalid argument IP address 0.0.0.111 is very strange and look like callerID 111 | ||||
| Comments: | By: Badalian Vyacheslav (slavon) 2011-09-09 08:42:10.266-0500 SIP Debugging Enabled for IP: 192.168.100.95 office*CLI> office*CLI> office*CLI> office*CLI> office*CLI> office*CLI> office*CLI> office*CLI> office*CLI> office*CLI> office*CLI> office*CLI> office*CLI> Reliably Transmitting (no NAT) to 192.168.100.95:5060: OPTIONS sip:111_office@192.168.100.95:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.100.1:5060;branch=z9hG4bK7cb36d9d Max-Forwards: 70 From: "asterisk" <sip:asterisk@192.168.100.1>;tag=as274f220b To: <sip:111_office@192.168.100.95:5060> Contact: <sip:asterisk@192.168.100.1:5060> Call-ID: 53206e8a1ed8b0f73c220b6d64defb26@192.168.100.1:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.6.0 Date: Fri, 09 Sep 2011 13:40:06 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- <--- SIP read from UDP:192.168.100.95:5060 ---> SIP/2.0 200 OK To: <sip:111_office@192.168.100.95:5060>;tag=98b4e89556f4dedei0 From: "asterisk" <sip:asterisk@192.168.100.1>;tag=as274f220b Call-ID: 53206e8a1ed8b0f73c220b6d64defb26@192.168.100.1:5060 CSeq: 102 OPTIONS Via: SIP/2.0/UDP 192.168.100.1:5060;branch=z9hG4bK7cb36d9d Server: Linksys/SPA941-5.1.8 Content-Length: 0 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: replaces <-------------> --- (10 headers 0 lines) --- Really destroying SIP dialog '53206e8a1ed8b0f73c220b6d64defb26@192.168.100.1:5060' Method: OPTIONS Reliably Transmitting (no NAT) to 192.168.100.95:5060: OPTIONS sip:111_office@192.168.100.95:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.100.1:5060;branch=z9hG4bK6857d5b2 Max-Forwards: 70 From: "asterisk" <sip:asterisk@192.168.100.1>;tag=as7fe114a6 To: <sip:111_office@192.168.100.95:5060> Contact: <sip:asterisk@192.168.100.1:5060> Call-ID: 1407e1eb0baebc8f587fa44079cc1a70@192.168.100.1:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.6.0 Date: Fri, 09 Sep 2011 13:40:11 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- <--- SIP read from UDP:192.168.100.95:5060 ---> SIP/2.0 200 OK To: <sip:111_office@192.168.100.95:5060>;tag=98b4e89556f4dedei0 From: "asterisk" <sip:asterisk@192.168.100.1>;tag=as7fe114a6 Call-ID: 1407e1eb0baebc8f587fa44079cc1a70@192.168.100.1:5060 CSeq: 102 OPTIONS Via: SIP/2.0/UDP 192.168.100.1:5060;branch=z9hG4bK6857d5b2 Server: Linksys/SPA941-5.1.8 Content-Length: 0 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: replaces <-------------> --- (10 headers 0 lines) --- Really destroying SIP dialog '1407e1eb0baebc8f587fa44079cc1a70@192.168.100.1:5060' Method: OPTIONS == Using SIP RTP CoS mark 5 -- Executing [74956660111@in:1] Verbose("SIP/a74956660111-000000e7", "9250102711 9250102711") in new stack 9250102711 9250102711 -- Executing [74956660111@in:2] Set("SIP/a74956660111-000000e7", "_x_callerid=9250102711") in new stack -- Executing [74956660111@in:3] Set("SIP/a74956660111-000000e7", "_x_direction=in") in new stack -- Executing [74956660111@in:4] Goto("SIP/a74956660111-000000e7", "IVR,s,1") in new stack -- Goto (IVR,s,1) -- Executing [s@IVR:1] BackGround("SIP/a74956660111-000000e7", "greetengs_hello") in new stack -- <SIP/a74956660111-000000e7> Playing 'greetengs_hello.slin' (language 'ru') Reliably Transmitting (no NAT) to 192.168.100.95:5060: OPTIONS sip:111_office@192.168.100.95:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.100.1:5060;branch=z9hG4bK591efa88 Max-Forwards: 70 From: "asterisk" <sip:asterisk@192.168.100.1>;tag=as10eedf9f To: <sip:111_office@192.168.100.95:5060> Contact: <sip:asterisk@192.168.100.1:5060> Call-ID: 189f58db08690de579b9c87b5be6fad0@192.168.100.1:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.6.0 Date: Fri, 09 Sep 2011 13:40:16 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- <--- SIP read from UDP:192.168.100.95:5060 ---> SIP/2.0 200 OK To: <sip:111_office@192.168.100.95:5060>;tag=98b4e89556f4dedei0 From: "asterisk" <sip:asterisk@192.168.100.1>;tag=as10eedf9f Call-ID: 189f58db08690de579b9c87b5be6fad0@192.168.100.1:5060 CSeq: 102 OPTIONS Via: SIP/2.0/UDP 192.168.100.1:5060;branch=z9hG4bK591efa88 Server: Linksys/SPA941-5.1.8 Content-Length: 0 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: replaces <-------------> --- (10 headers 0 lines) --- Really destroying SIP dialog '189f58db08690de579b9c87b5be6fad0@192.168.100.1:5060' Method: OPTIONS -- Executing [s@IVR:2] BackGround("SIP/a74956660111-000000e7", "greetengs_to_connect") in new stack -- <SIP/a74956660111-000000e7> Playing 'greetengs_to_connect.slin' (language 'ru') Reliably Transmitting (no NAT) to 192.168.100.95:5060: OPTIONS sip:111_office@192.168.100.95:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.100.1:5060;branch=z9hG4bK6e4b75a7 Max-Forwards: 70 From: "asterisk" <sip:asterisk@192.168.100.1>;tag=as4e91fbab To: <sip:111_office@192.168.100.95:5060> Contact: <sip:asterisk@192.168.100.1:5060> Call-ID: 72727b1401cea7d62bd3d6dc07167273@192.168.100.1:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.6.0 Date: Fri, 09 Sep 2011 13:40:21 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- <--- SIP read from UDP:192.168.100.95:5060 ---> SIP/2.0 200 OK To: <sip:111_office@192.168.100.95:5060>;tag=98b4e89556f4dedei0 From: "asterisk" <sip:asterisk@192.168.100.1>;tag=as4e91fbab Call-ID: 72727b1401cea7d62bd3d6dc07167273@192.168.100.1:5060 CSeq: 102 OPTIONS Via: SIP/2.0/UDP 192.168.100.1:5060;branch=z9hG4bK6e4b75a7 Server: Linksys/SPA941-5.1.8 Content-Length: 0 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: replaces <-------------> --- (10 headers 0 lines) --- Really destroying SIP dialog '72727b1401cea7d62bd3d6dc07167273@192.168.100.1:5060' Method: OPTIONS == CDR updated on SIP/a74956660111-000000e7 -- Executing [111@IVR:1] Dial("SIP/a74956660111-000000e7", "SIP/111_office&SIP/111_pc&SIP/111_notebook&SIP/111_mobile&Local/111@office_mobile_redirect,,mt") in new stack == Using SIP RTP CoS mark 5 Audio is at 5060 Video is at 192.168.100.1:5060 Adding codec 0x8 (alaw) to SDP Adding codec 0x4 (ulaw) to SDP Adding video codec 0x40000 (h261) to SDP Adding video codec 0x80000 (h263) to SDP Adding video codec 0x100000 (h263p) to SDP Adding video codec 0x200000 (h264) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 192.168.100.95:5060: INVITE sip:111_office@192.168.100.95:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.100.1:5060;branch=z9hG4bK3df05521 Max-Forwards: 70 From: "9250102711" <sip:9250102711@192.168.100.1>;tag=as1a8eb74e To: <sip:111_office@192.168.100.95:5060> Contact: <sip:9250102711@192.168.100.1:5060> Call-ID: 2eb6f4002b40aca86522651744a8d253@192.168.100.1:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.8.6.0 Date: Fri, 09 Sep 2011 13:40:22 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Remote-Party-ID: "9250102711" <sip:9250102711@192.168.100.1>;party=calling;privacy=off;screen=no Content-Type: application/sdp Content-Length: 418 v=0 o=root 249624956 249624956 IN IP4 192.168.100.1 s=Asterisk PBX 1.8.6.0 c=IN IP4 192.168.100.1 b=CT:384 t=0 0 m=audio 16028 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv m=video 15086 RTP/AVP 31 34 98 99 a=rtpmap:31 H261/90000 a=rtpmap:34 H263/90000 a=rtpmap:98 h263-1998/90000 a=rtpmap:99 H264/90000 a=sendrecv --- -- Called SIP/111_office [Sep 9 17:40:22] WARNING[21582]: app_dial.c:2197 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown) [Sep 9 17:40:22] WARNING[21582]: app_dial.c:2197 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown) [Sep 9 17:40:22] WARNING[21582]: app_dial.c:2197 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown) [Sep 9 17:40:22] NOTICE[21582]: chan_local.c:899 local_call: No such extension/context 111@office_mobile_redirect while calling Local channel -- Couldn't call Local/111@office_mobile_redirect -- Started music on hold, class 'default', on SIP/a74956660111-000000e7 <--- SIP read from UDP:192.168.100.95:5060 ---> SIP/2.0 100 Trying To: <sip:111_office@192.168.100.95:5060> From: "9250102711" <sip:9250102711@192.168.100.1>;tag=as1a8eb74e Call-ID: 2eb6f4002b40aca86522651744a8d253@192.168.100.1:5060 CSeq: 102 INVITE Via: SIP/2.0/UDP 192.168.100.1:5060;branch=z9hG4bK3df05521 Server: Linksys/SPA941-5.1.8 Content-Length: 0 <-------------> --- (8 headers 0 lines) --- <--- SIP read from UDP:192.168.100.95:5060 ---> SIP/2.0 180 Ringing To: <sip:111_office@192.168.100.95:5060>;tag=41178b994afc3c3ai0 From: "9250102711" <sip:9250102711@192.168.100.1>;tag=as1a8eb74e Call-ID: 2eb6f4002b40aca86522651744a8d253@192.168.100.1:5060 CSeq: 102 INVITE Via: SIP/2.0/UDP 192.168.100.1:5060;branch=z9hG4bK3df05521 Server: Linksys/SPA941-5.1.8 Content-Length: 0 <-------------> --- (8 headers 0 lines) --- -- SIP/111_office-000000e8 is ringing <--- SIP read from UDP:192.168.100.95:5060 ---> SIP/2.0 200 OK To: <sip:111_office@192.168.100.95:5060>;tag=41178b994afc3c3ai0 From: "9250102711" <sip:9250102711@192.168.100.1>;tag=as1a8eb74e Call-ID: 2eb6f4002b40aca86522651744a8d253@192.168.100.1:5060 CSeq: 102 INVITE Via: SIP/2.0/UDP 192.168.100.1:5060;branch=z9hG4bK3df05521 Contact: "Alexandr Sidorov <111>" <sip:111_office@192.168.100.95:5060> Server: Linksys/SPA941-5.1.8 Content-Length: 243 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: replaces Content-Type: application/sdp v=0 o=- 9652352 9652352 IN IP4 192.168.100.95 s=- c=IN IP4 192.168.100.95 t=0 0 m=audio 16448 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv m=video 0 RTP/AVP 31 34 98 99 <-------------> --- (12 headers 12 lines) --- Found RTP audio format 0 Found RTP audio format 101 Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 [Sep 9 17:40:24] WARNING[11319]: chan_sip.c:8738 process_sdp: Unsupported SDP media type in offer: video 0 RTP/AVP 31 34 98 99 Capabilities: us - 0x3c000c (ulaw|alaw|h261|h263|h263p|h264), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 192.168.100.95:16448 Peer doesn't provide video list_route: hop: <111> set_destination: Parsing <111> for address/port to send to set_destination: set destination to 0.0.0.111:5060 [Sep 9 17:40:24] WARNING[11319]: chan_sip.c:3349 __sip_xmit: sip_xmit of 0x1bae2d30 (len 404) to 0.0.0.111:5060 returned -1: Invalid argument -- SIP/111_office-000000e8 answered SIP/a74956660111-000000e7 -- Stopped music on hold on SIP/a74956660111-000000e7 <--- SIP read from UDP:192.168.100.95:5060 ---> SIP/2.0 200 OK To: <sip:111_office@192.168.100.95:5060>;tag=41178b994afc3c3ai0 From: "9250102711" <sip:9250102711@192.168.100.1>;tag=as1a8eb74e Call-ID: 2eb6f4002b40aca86522651744a8d253@192.168.100.1:5060 CSeq: 102 INVITE Via: SIP/2.0/UDP 192.168.100.1:5060;branch=z9hG4bK3df05521 Contact: "Alexandr Sidorov <111>" <sip:111_office@192.168.100.95:5060> Server: Linksys/SPA941-5.1.8 Content-Length: 243 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: replaces Content-Type: application/sdp v=0 o=- 9652352 9652352 IN IP4 192.168.100.95 s=- c=IN IP4 192.168.100.95 t=0 0 m=audio 16448 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv m=video 0 RTP/AVP 31 34 98 99 <-------------> --- (12 headers 12 lines) --- [Sep 9 17:40:24] WARNING[11319]: chan_sip.c:3349 __sip_xmit: sip_xmit of 0x1bb87530 (len 404) to 0.0.0.111:5060 returned -1: Invalid argument <--- SIP read from UDP:192.168.100.95:5060 ---> SIP/2.0 200 OK To: <sip:111_office@192.168.100.95:5060>;tag=41178b994afc3c3ai0 From: "9250102711" <sip:9250102711@192.168.100.1>;tag=as1a8eb74e Call-ID: 2eb6f4002b40aca86522651744a8d253@192.168.100.1:5060 CSeq: 102 INVITE Via: SIP/2.0/UDP 192.168.100.1:5060;branch=z9hG4bK3df05521 Contact: "Alexandr Sidorov <111>" <sip:111_office@192.168.100.95:5060> Server: Linksys/SPA941-5.1.8 Content-Length: 243 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: replaces Content-Type: application/sdp v=0 o=- 9652352 9652352 IN IP4 192.168.100.95 s=- c=IN IP4 192.168.100.95 t=0 0 m=audio 16448 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv m=video 0 RTP/AVP 31 34 98 99 <-------------> --- (12 headers 12 lines) --- [Sep 9 17:40:25] WARNING[11319]: chan_sip.c:3349 __sip_xmit: sip_xmit of 0x1bb87530 (len 404) to 0.0.0.111:5060 returned -1: Invalid argument Reliably Transmitting (no NAT) to 192.168.100.95:5060: OPTIONS sip:111_office@192.168.100.95:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.100.1:5060;branch=z9hG4bK6a05afb0 Max-Forwards: 70 From: "asterisk" <sip:asterisk@192.168.100.1>;tag=as3bc72aa9 To: <sip:111_office@192.168.100.95:5060> Contact: <sip:asterisk@192.168.100.1:5060> Call-ID: 77226998130275ad74750f7c66974505@192.168.100.1:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.6.0 Date: Fri, 09 Sep 2011 13:40:26 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- <--- SIP read from UDP:192.168.100.95:5060 ---> SIP/2.0 200 OK To: <sip:111_office@192.168.100.95:5060>;tag=98b4e89556f4dedei0 From: "asterisk" <sip:asterisk@192.168.100.1>;tag=as3bc72aa9 Call-ID: 77226998130275ad74750f7c66974505@192.168.100.1:5060 CSeq: 102 OPTIONS Via: SIP/2.0/UDP 192.168.100.1:5060;branch=z9hG4bK6a05afb0 Server: Linksys/SPA941-5.1.8 Content-Length: 0 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: replaces <-------------> --- (10 headers 0 lines) --- Really destroying SIP dialog '77226998130275ad74750f7c66974505@192.168.100.1:5060' Method: OPTIONS <--- SIP read from UDP:192.168.100.95:5060 ---> SIP/2.0 200 OK To: <sip:111_office@192.168.100.95:5060>;tag=41178b994afc3c3ai0 From: "9250102711" <sip:9250102711@192.168.100.1>;tag=as1a8eb74e Call-ID: 2eb6f4002b40aca86522651744a8d253@192.168.100.1:5060 CSeq: 102 INVITE Via: SIP/2.0/UDP 192.168.100.1:5060;branch=z9hG4bK3df05521 Contact: "Alexandr Sidorov <111>" <sip:111_office@192.168.100.95:5060> Server: Linksys/SPA941-5.1.8 Content-Length: 243 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: replaces Content-Type: application/sdp v=0 o=- 9652352 9652352 IN IP4 192.168.100.95 s=- c=IN IP4 192.168.100.95 t=0 0 m=audio 16448 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv m=video 0 RTP/AVP 31 34 98 99 <-------------> --- (12 headers 12 lines) --- [Sep 9 17:40:27] WARNING[11319]: chan_sip.c:3349 __sip_xmit: sip_xmit of 0x1bb87530 (len 404) to 0.0.0.111:5060 returned -1: Invalid argument [Sep 9 17:40:28] NOTICE[11319]: chan_sip.c:12593 sip_reregister: -- Re-registration for a74956660111@87.255.0.218 [Sep 9 17:40:28] NOTICE[11319]: chan_sip.c:20125 handle_response_register: Outbound Registration: Expiry for 87.255.0.218 is 120 sec (Scheduling reregistration in 105 s) Reliably Transmitting (no NAT) to 192.168.100.95:5060: OPTIONS sip:111_office@192.168.100.95:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.100.1:5060;branch=z9hG4bK78fb0901 Max-Forwards: 70 From: "asterisk" <sip:asterisk@192.168.100.1>;tag=as47640e14 To: <sip:111_office@192.168.100.95:5060> Contact: <sip:asterisk@192.168.100.1:5060> Call-ID: 65991fc53562497e199cd9bb350f5695@192.168.100.1:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.6.0 Date: Fri, 09 Sep 2011 13:40:31 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- <--- SIP read from UDP:192.168.100.95:5060 ---> SIP/2.0 200 OK To: <sip:111_office@192.168.100.95:5060>;tag=98b4e89556f4dedei0 From: "asterisk" <sip:asterisk@192.168.100.1>;tag=as47640e14 Call-ID: 65991fc53562497e199cd9bb350f5695@192.168.100.1:5060 CSeq: 102 OPTIONS Via: SIP/2.0/UDP 192.168.100.1:5060;branch=z9hG4bK78fb0901 Server: Linksys/SPA941-5.1.8 Content-Length: 0 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: replaces <-------------> --- (10 headers 0 lines) --- Really destroying SIP dialog '65991fc53562497e199cd9bb350f5695@192.168.100.1:5060' Method: OPTIONS <--- SIP read from UDP:192.168.100.95:5060 ---> SIP/2.0 200 OK To: <sip:111_office@192.168.100.95:5060>;tag=41178b994afc3c3ai0 From: "9250102711" <sip:9250102711@192.168.100.1>;tag=as1a8eb74e Call-ID: 2eb6f4002b40aca86522651744a8d253@192.168.100.1:5060 CSeq: 102 INVITE Via: SIP/2.0/UDP 192.168.100.1:5060;branch=z9hG4bK3df05521 Contact: "Alexandr Sidorov <111>" <sip:111_office@192.168.100.95:5060> Server: Linksys/SPA941-5.1.8 Content-Length: 243 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: replaces Content-Type: application/sdp v=0 o=- 9652352 9652352 IN IP4 192.168.100.95 s=- c=IN IP4 192.168.100.95 t=0 0 m=audio 16448 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv m=video 0 RTP/AVP 31 34 98 99 <-------------> --- (12 headers 12 lines) --- [Sep 9 17:40:31] WARNING[11319]: chan_sip.c:3349 __sip_xmit: sip_xmit of 0x1bb87530 (len 404) to 0.0.0.111:5060 returned -1: Invalid argument <--- SIP read from UDP:192.168.100.95:5060 ---> SIP/2.0 200 OK To: <sip:111_office@192.168.100.95:5060>;tag=41178b994afc3c3ai0 From: "9250102711" <sip:9250102711@192.168.100.1>;tag=as1a8eb74e Call-ID: 2eb6f4002b40aca86522651744a8d253@192.168.100.1:5060 CSeq: 102 INVITE Via: SIP/2.0/UDP 192.168.100.1:5060;branch=z9hG4bK3df05521 Contact: "Alexandr Sidorov <111>" <sip:111_office@192.168.100.95:5060> Server: Linksys/SPA941-5.1.8 Content-Length: 243 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: replaces Content-Type: application/sdp v=0 o=- 9652352 9652352 IN IP4 192.168.100.95 s=- c=IN IP4 192.168.100.95 t=0 0 m=audio 16448 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv m=video 0 RTP/AVP 31 34 98 99 <-------------> --- (12 headers 12 lines) --- [Sep 9 17:40:35] WARNING[11319]: chan_sip.c:3349 __sip_xmit: sip_xmit of 0x2aaaac126d90 (len 404) to 0.0.0.111:5060 returned -1: Invalid argument -- Executing [h@IVR:1] Set("SIP/a74956660111-000000e7", "CDR(userfield)=in:9250102711:9250102711:111:11:ANSWERED") in new stack [Sep 9 17:40:36] WARNING[21582]: chan_sip.c:3349 __sip_xmit: sip_xmit of 0x1bb61fc0 (len 431) to 0.0.0.111:5060 returned -1: Invalid argument == Spawn extension (IVR, 111, 1) exited non-zero on 'SIP/a74956660111-000000e7' [Sep 9 17:40:36] WARNING[11319]: chan_sip.c:3349 __sip_xmit: sip_xmit of 0x1bb61fc0 (len 431) to 0.0.0.111:5060 returned -1: Invalid argument [Sep 9 17:40:36] WARNING[11319]: chan_sip.c:3349 __sip_xmit: sip_xmit of 0x1bb61fc0 (len 431) to 0.0.0.111:5060 returned -1: Invalid argument Reliably Transmitting (no NAT) to 192.168.100.95:5060: OPTIONS sip:111_office@192.168.100.95:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.100.1:5060;branch=z9hG4bK3649ae11 Max-Forwards: 70 From: "asterisk" <sip:asterisk@192.168.100.1>;tag=as69d437f2 To: <sip:111_office@192.168.100.95:5060> Contact: <sip:asterisk@192.168.100.1:5060> Call-ID: 5d9e42b8581969124a2a95825827ad34@192.168.100.1:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.6.0 Date: Fri, 09 Sep 2011 13:40:36 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- <--- SIP read from UDP:192.168.100.95:5060 ---> SIP/2.0 200 OK To: <sip:111_office@192.168.100.95:5060>;tag=98b4e89556f4dedei0 From: "asterisk" <sip:asterisk@192.168.100.1>;tag=as69d437f2 Call-ID: 5d9e42b8581969124a2a95825827ad34@192.168.100.1:5060 CSeq: 102 OPTIONS Via: SIP/2.0/UDP 192.168.100.1:5060;branch=z9hG4bK3649ae11 Server: Linksys/SPA941-5.1.8 Content-Length: 0 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: replaces <-------------> --- (10 headers 0 lines) --- Really destroying SIP dialog '5d9e42b8581969124a2a95825827ad34@192.168.100.1:5060' Method: OPTIONS [Sep 9 17:40:37] WARNING[11319]: chan_sip.c:3349 __sip_xmit: sip_xmit of 0x1bb61fc0 (len 431) to 0.0.0.111:5060 returned -1: Invalid argument [Sep 9 17:40:37] WARNING[11319]: chan_sip.c:3349 __sip_xmit: sip_xmit of 0x1bb61fc0 (len 431) to 0.0.0.111:5060 returned -1: Invalid argument [Sep 9 17:40:39] WARNING[11319]: chan_sip.c:3349 __sip_xmit: sip_xmit of 0x1bb61fc0 (len 431) to 0.0.0.111:5060 returned -1: Invalid argument <--- SIP read from UDP:192.168.100.95:5060 ---> SIP/2.0 200 OK To: <sip:111_office@192.168.100.95:5060>;tag=41178b994afc3c3ai0 From: "9250102711" <sip:9250102711@192.168.100.1>;tag=as1a8eb74e Call-ID: 2eb6f4002b40aca86522651744a8d253@192.168.100.1:5060 CSeq: 102 INVITE Via: SIP/2.0/UDP 192.168.100.1:5060;branch=z9hG4bK3df05521 Contact: "Alexandr Sidorov <111>" <sip:111_office@192.168.100.95:5060> Server: Linksys/SPA941-5.1.8 Content-Length: 243 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: replaces Content-Type: application/sdp v=0 o=- 9652352 9652352 IN IP4 192.168.100.95 s=- c=IN IP4 192.168.100.95 t=0 0 m=audio 16448 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv m=video 0 RTP/AVP 31 34 98 99 <-------------> --- (12 headers 12 lines) --- [Sep 9 17:40:39] WARNING[11319]: chan_sip.c:3349 __sip_xmit: sip_xmit of 0x2aaaac126d90 (len 404) to 0.0.0.111:5060 returned -1: Invalid argument [Sep 9 17:40:40] NOTICE[11319]: chan_sip.c:12593 sip_reregister: -- Re-registration for open.bs@sipnet.ru [Sep 9 17:40:41] NOTICE[11319]: chan_sip.c:20125 handle_response_register: Outbound Registration: Expiry for sipnet.ru is 118 sec (Scheduling reregistration in 103 s) Reliably Transmitting (no NAT) to 192.168.100.95:5060: OPTIONS sip:111_office@192.168.100.95:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.100.1:5060;branch=z9hG4bK303e15b5 Max-Forwards: 70 From: "asterisk" <sip:asterisk@192.168.100.1>;tag=as2fabff2b To: <sip:111_office@192.168.100.95:5060> Contact: <sip:asterisk@192.168.100.1:5060> Call-ID: 232bec3349a8339c75802c433d1ddb2a@192.168.100.1:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.6.0 Date: Fri, 09 Sep 2011 13:40:41 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- <--- SIP read from UDP:192.168.100.95:5060 ---> SIP/2.0 200 OK To: <sip:111_office@192.168.100.95:5060>;tag=98b4e89556f4dedei0 From: "asterisk" <sip:asterisk@192.168.100.1>;tag=as2fabff2b Call-ID: 232bec3349a8339c75802c433d1ddb2a@192.168.100.1:5060 CSeq: 102 OPTIONS Via: SIP/2.0/UDP 192.168.100.1:5060;branch=z9hG4bK303e15b5 Server: Linksys/SPA941-5.1.8 Content-Length: 0 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: replaces <-------------> --- (10 headers 0 lines) --- Really destroying SIP dialog '232bec3349a8339c75802c433d1ddb2a@192.168.100.1:5060' Method: OPTIONS [Sep 9 17:40:42] WARNING[11319]: chan_sip.c:3349 __sip_xmit: sip_xmit of 0x1bb61fc0 (len 431) to 0.0.0.111:5060 returned -1: Invalid argument [Sep 9 17:40:42] WARNING[11319]: chan_sip.c:3620 retrans_pkt: Retransmission timeout reached on transmission 2eb6f4002b40aca86522651744a8d253@192.168.100.1:5060 for seqno 103 (Non-critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 6400ms with no response Reliably Transmitting (no NAT) to 192.168.100.95:5060: OPTIONS sip:111_office@192.168.100.95:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.100.1:5060;branch=z9hG4bK6190ef59 Max-Forwards: 70 From: "asterisk" <sip:asterisk@192.168.100.1>;tag=as19f3a34c To: <sip:111_office@192.168.100.95:5060> Contact: <sip:asterisk@192.168.100.1:5060> Call-ID: 110d83767be485e91082ac056bff05be@192.168.100.1:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.6.0 Date: Fri, 09 Sep 2011 13:40:46 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- <--- SIP read from UDP:192.168.100.95:5060 ---> SIP/2.0 200 OK To: <sip:111_office@192.168.100.95:5060>;tag=98b4e89556f4dedei0 From: "asterisk" <sip:asterisk@192.168.100.1>;tag=as19f3a34c Call-ID: 110d83767be485e91082ac056bff05be@192.168.100.1:5060 CSeq: 102 OPTIONS Via: SIP/2.0/UDP 192.168.100.1:5060;branch=z9hG4bK6190ef59 Server: Linksys/SPA941-5.1.8 Content-Length: 0 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: replaces <-------------> --- (10 headers 0 lines) --- Really destroying SIP dialog '110d83767be485e91082ac056bff05be@192.168.100.1:5060' Method: OPTIONS By: Leif Madsen (lmadsen) 2011-09-12 12:52:37.314-0500 Duplicate issue. By: Badalian Vyacheslav (slavon) 2011-09-16 08:24:38.576-0500 Sorry but is no duplicate issue! Phone is registred! office*CLI> sip show peers Name/username Host Dyn Forcerport ACL Port Status 111_office/111_office 192.168.100.95 D 5060 OK (20 ms) Call from phone is good. Two side sound Call to phone is bad - one side sound. In Asterisk 1.6 all is work normal. After upgrade to 1.8 we have bug. | ||||