Re: [asterisk-users] WARNING chan_sip.c:3115 __sip_xmit
I ran tcpdump on version 1.6 and 1.8 and compare sip header and i found in 1.8 asterisk if you call non-exiting peer/exten its waiting for ACK packet for 100 Tying message and in 1.6 its not that why i am getting following messages __sip_xmit: sip_xmit blah..blah See following header of sip 1.8 its almost waiting 45 sec to get ACK packet.. and declarer peer not exist why this is not happen with 1.2, 1.4, 1.6 version ? 12:38:46.704472 IP (tos 0x0, ttl 64, id 0, offset 0, flags [DF], proto UDP (17), length 623) dhcp-254-211.east.ora.com.sip satish-desktop.sip: SIP, length: 595 INVITE sip:7103@172.30.245.208:5060 SIP/2.0 Via: SIP/2.0/UDP 172.30.254.211:5060;branch=z9hG4bK-3036-1-0 From: sipp sip:sipp@172.30.254.211:5060;tag=3036SIPpTag091 To: sut sip:7103@172.30.245.208:5060 Call-ID: 1-3036@172.30.254.211 CSeq: 1 INVITE Contact: sip:sipp@172.30.254.211:5060 Max-Forwards: 70 Subject: Performance Test Content-Type: application/sdp Content-Length: 198 v=0 o=user1 53655765 2353687637 IN IP4 172.30.254.211 s=- c=IN IP4 172.30.254.211 t=0 0 m=audio 6000 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11,16 12:38:46.705445 IP (tos 0x0, ttl 64, id 1416, offset 0, flags [none], proto UDP (17), length 487) satish-desktop.sip dhcp-254-211.east.ora.com.sip: SIP, length: 459 SIP/2.0 100 Trying Via: SIP/2.0/UDP 172.30.254.211:5060;branch=z9hG4bK-3036-1-0;received=172.30.254.211 From: sipp sip:sipp@172.30.254.211:5060;tag=3036SIPpTag091 To: sut sip:7103@172.30.245.208:5060 Call-ID: 1-3036@172.30.254.211 CSeq: 1 INVITE Server: Asterisk PBX 1.8.3.2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: sip:7103@172.30.245.208:5060 Content-Length: 0 12:39:18.706783 IP (tos 0x0, ttl 64, id 1417, offset 0, flags [none], proto UDP (17), length 771) satish-desktop.sip dhcp-254-211.east.ora.com.sip: SIP, length: 743 SIP/2.0 200 OK Via: SIP/2.0/UDP 172.30.254.211:5060;branch=z9hG4bK-3036-1-0;received=172.30.254.211 From: sipp sip:sipp@172.30.254.211:5060;tag=3036SIPpTag091 To: sut sip:7103@172.30.245.208:5060;tag=as7403b6f3 Call-ID: 1-3036@172.30.254.211 CSeq: 1 INVITE Server: Asterisk PBX 1.8.3.2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: sip:7103@172.30.245.208:5060 Content-Type: application/sdp Content-Length: 240 v=0 o=root 1076282210 1076282210 IN IP4 172.30.245.208 s=Asterisk PBX 1.8.3.2 c=IN IP4 172.30.245.208 t=0 0 m=audio 17450 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv Date: Thu, 7 Apr 2011 16:40:12 -0400 From: p...@dugasenterprises.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] WARNING chan_sip.c:3115 __sip_xmit Just a guess but is it possible one of your SIP peers (7623 or 7624) has an invalid IP address of 0.0.29.200? I wonder what sip show peers shows. On Thu, Apr 7, 2011 at 4:03 PM, satish patel satish...@hotmail.com wrote: Re-opening this issue. If i dial number which doesn't existing then i am getting following error. So is there anyway i can fix my dialplan to check whether that number exist or not or i can check channel status. shirley*CLI == Using SIP RTP CoS mark 5 -- Executing [7623@from-sip:1] Macro(SIP/7527-0032, stdexten,7623,sip/7623sip/7624) in new stack -- Executing [s@macro-stdexten:1] Dial(SIP/7527-0032, sip/7623sip/7624IAX2/7623,20,t) in new stack [Apr 7 15:58:34] WARNING[14006]: app_dial.c:2039 dial_exec_full: Unable to create channel of type 'sip' (cause 20 - Unknown) == Using SIP RTP CoS mark 5 [Apr 7 15:58:34] WARNING[14006]: acl.c:698 ast_ouraddrfor: Cannot connect [Apr 7 15:58:34] WARNING[14006]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x2e08d50 (len 789) to 0.0.29.200:5060 returned -1: Invalid argument -- Called 7624 -- Called 7623 [Apr 7 15:58:34] WARNING[13920]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x2e08d50 (len 789) to 0.0.29.200:5060 returned -1: Invalid argument [Apr 7 15:58:35] WARNING[13920]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x2e08d50 (len 789) to 0.0.29.200:5060 returned -1: Invalid argument [Apr 7 15:58:37] WARNING[13920]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x2e08d50 (len 789) to 0.0.29.200:5060 returned -1: Invalid argument [Apr 7 15:58:38] NOTICE[13914]: chan_iax2.c:4643 __auto_congest: Auto-congesting call due to slow response
Re: [asterisk-users] WARNING chan_sip.c:3115 __sip_xmit
Re-opening this issue. If i dial number which doesn't existing then i am getting following error. So is there anyway i can fix my dialplan to check whether that number exist or not or i can check channel status. shirley*CLI == Using SIP RTP CoS mark 5 -- Executing [7623@from-sip:1] Macro(SIP/7527-0032, stdexten,7623,sip/7623sip/7624) in new stack -- Executing [s@macro-stdexten:1] Dial(SIP/7527-0032, sip/7623sip/7624IAX2/7623,20,t) in new stack [Apr 7 15:58:34] WARNING[14006]: app_dial.c:2039 dial_exec_full: Unable to create channel of type 'sip' (cause 20 - Unknown) == Using SIP RTP CoS mark 5 [Apr 7 15:58:34] WARNING[14006]: acl.c:698 ast_ouraddrfor: Cannot connect [Apr 7 15:58:34] WARNING[14006]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x2e08d50 (len 789) to 0.0.29.200:5060 returned -1: Invalid argument -- Called 7624 -- Called 7623 [Apr 7 15:58:34] WARNING[13920]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x2e08d50 (len 789) to 0.0.29.200:5060 returned -1: Invalid argument [Apr 7 15:58:35] WARNING[13920]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x2e08d50 (len 789) to 0.0.29.200:5060 returned -1: Invalid argument [Apr 7 15:58:37] WARNING[13920]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x2e08d50 (len 789) to 0.0.29.200:5060 returned -1: Invalid argument [Apr 7 15:58:38] NOTICE[13914]: chan_iax2.c:4643 __auto_congest: Auto-congesting call due to slow response -- IAX2/0.0.29.199:4569-13525 is circuit-busy -- Hungup 'IAX2/0.0.29.199:4569-13525' [Apr 7 15:58:39] WARNING[13920]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x7fd08e3f45a0 (len 787) to 0.0.29.200:5060 returned -1: Invalid argument [Apr 7 15:58:40] WARNING[13920]: chan_sip.c:3386 retrans_pkt: Retransmission timeout reached on transmission 6cf13d63561e7c106c31ffb74571e661@172.30.1.47:5060 for seqno 102 (Critical Request) -- See doc/sip-retransmit.txt. Packet timed out after 32000ms with no response [Apr 7 15:58:41] WARNING[13920]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x2e08d50 (len 789) to 0.0.29.200:5060 returned -1: Invalid argument == Spawn extension (macro-stdexten, s, 1) exited non-zero on 'SIP/7527-0032' in macro 'stdexten' == Spawn extension (from-sip, 7623, 1) exited non-zero on 'SIP/7527-0032' [Apr 7 15:58:49] WARNING[13920]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x2e08d50 (len 789) to 0.0.29.200:5060 returned -1: Invalid argument From: satish...@hotmail.com To: asterisk-users@lists.digium.com Date: Mon, 4 Apr 2011 20:22:55 + Subject: Re: [asterisk-users] WARNING chan_sip.c:3115 __sip_xmit Thanks for reply! I found this problem only with X-lite version of softphone. Other phones are working fine without any WARNING! look like X-lite has some short of SIP issue. -S From: mden...@gmail.com Date: Mon, 4 Apr 2011 15:59:43 -0400 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] WARNING chan_sip.c:3115 __sip_xmit On Mon, Apr 4, 2011 at 3:51 PM, satish patel satish...@hotmail.com wrote: Hey Guys, Whenever i calling any extension i am getting following WARNING messages do you have any idea they coming from where? -Satish shirley*CLI == Using SIP RTP CoS mark 5 -- Executing [7623@from-sip:1] Macro(SIP/7527-0008, stdexten,7623,sip/7623sip/7624) in new stack -- Executing [s@macro-stdexten:1] Dial(SIP/7527-0008, sip/7623sip/7624iax2/7623,20,t) in new stack == Using SIP RTP CoS mark 5 -- Called 7623 == Using SIP RTP CoS mark 5 [Apr 4 12:46:37] WARNING[6003]: acl.c:698 ast_ouraddrfor: Cannot connect [Apr 4 12:46:37] WARNING[6003]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x8379240 (len 787) to 0.0.29.200:5060 returned -1: Invalid argument -- Called 7624 -- Called 7623 -- SIP/7623-0009 is ringing [Apr 4 12:46:38] WARNING[5982]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x8379240 (len 787) to 0.0.29.200:5060 returned -1: Invalid argument [Apr 4 12:46:39] WARNING[5982]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x8379240 (len 787) to 0.0.29.200:5060 returned -1: Invalid argument [Apr 4 12:46:41] WARNING[5982]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x8379240 (len 787) to 0.0.29.200:5060 returned -1: Invalid argument [Apr 4 12:46:41] NOTICE[5975]: chan_iax2.c:4643 __auto_congest: Auto-congesting call due to slow response -- IAX2/0.0.29.199:4569-5537 is circuit-busy -- Hungup 'IAX2/0.0.29.199:4569-5537' [Apr 4 12:46:45] WARNING[5982]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x8379240 (len 787) to 0.0.29.200:5060 returned -1: Invalid argument -- SIP/7623-0009 connected line has changed. Saving it until answer for SIP/7527-0008 -- SIP/7623-0009 answered SIP/7527-0008 [Apr 4 12:46:53] WARNING[5982]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x8379240 (len 787) to 0.0.29.200:5060 returned -1: Invalid argument == Spawn extension (macro-stdexten, s, 1) exited non-zero
Re: [asterisk-users] WARNING chan_sip.c:3115 __sip_xmit
Just a guess but is it possible one of your SIP peers (7623 or 7624) has an invalid IP address of 0.0.29.200? I wonder what sip show peers shows. On Thu, Apr 7, 2011 at 4:03 PM, satish patel satish...@hotmail.com wrote: Re-opening this issue. If i dial number which doesn't existing then i am getting following error. So is there anyway i can fix my dialplan to check whether that number exist or not or i can check channel status. shirley*CLI == Using SIP RTP CoS mark 5 -- Executing [7623@from-sip:1] Macro(SIP/7527-0032, stdexten,7623,sip/7623sip/7624) in new stack -- Executing [s@macro-stdexten:1] Dial(SIP/7527-0032, sip/7623sip/7624IAX2/7623,20,t) in new stack [Apr 7 15:58:34] WARNING[14006]: app_dial.c:2039 dial_exec_full: Unable to create channel of type 'sip' (cause 20 - Unknown) == Using SIP RTP CoS mark 5 [Apr 7 15:58:34] WARNING[14006]: acl.c:698 ast_ouraddrfor: Cannot connect [Apr 7 15:58:34] WARNING[14006]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x2e08d50 (len 789) to 0.0.29.200:5060 returned -1: Invalid argument -- Called 7624 -- Called 7623 [Apr 7 15:58:34] WARNING[13920]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x2e08d50 (len 789) to 0.0.29.200:5060 returned -1: Invalid argument [Apr 7 15:58:35] WARNING[13920]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x2e08d50 (len 789) to 0.0.29.200:5060 returned -1: Invalid argument [Apr 7 15:58:37] WARNING[13920]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x2e08d50 (len 789) to 0.0.29.200:5060 returned -1: Invalid argument [Apr 7 15:58:38] NOTICE[13914]: chan_iax2.c:4643 __auto_congest: Auto-congesting call due to slow response -- IAX2/0.0.29.199:4569-13525 is circuit-busy -- Hungup 'IAX2/0.0.29.199:4569-13525' [Apr 7 15:58:39] WARNING[13920]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x7fd08e3f45a0 (len 787) to 0.0.29.200:5060 returned -1: Invalid argument [Apr 7 15:58:40] WARNING[13920]: chan_sip.c:3386 retrans_pkt: Retransmission timeout reached on transmission 6cf13d63561e7c106c31ffb74571e661@172.30.1.47:5060 for seqno 102 (Critical Request) -- See doc/sip-retransmit.txt. Packet timed out after 32000ms with no response [Apr 7 15:58:41] WARNING[13920]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x2e08d50 (len 789) to 0.0.29.200:5060 returned -1: Invalid argument == Spawn extension (macro-stdexten, s, 1) exited non-zero on 'SIP/7527-0032' in macro 'stdexten' == Spawn extension (from-sip, 7623, 1) exited non-zero on 'SIP/7527-0032' [Apr 7 15:58:49] WARNING[13920]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x2e08d50 (len 789) to 0.0.29.200:5060 returned -1: Invalid argument From: satish...@hotmail.com To: asterisk-users@lists.digium.com Date: Mon, 4 Apr 2011 20:22:55 + Subject: Re: [asterisk-users] WARNING chan_sip.c:3115 __sip_xmit Thanks for reply! I found this problem only with X-lite version of softphone. Other phones are working fine without any WARNING! look like X-lite has some short of SIP issue. -S From: mden...@gmail.com Date: Mon, 4 Apr 2011 15:59:43 -0400 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] WARNING chan_sip.c:3115 __sip_xmit On Mon, Apr 4, 2011 at 3:51 PM, satish patel satish...@hotmail.com wrote: Hey Guys, Whenever i calling any extension i am getting following WARNING messages do you have any idea they coming from where? -Satish shirley*CLI == Using SIP RTP CoS mark 5 -- Executing [7623@from-sip:1] Macro(SIP/7527-0008, stdexten,7623,sip/7623sip/7624) in new stack -- Executing [s@macro-stdexten:1] Dial(SIP/7527-0008, sip/7623sip/7624iax2/7623,20,t) in new stack == Using SIP RTP CoS mark 5 -- Called 7623 == Using SIP RTP CoS mark 5 [Apr 4 12:46:37] WARNING[6003]: acl.c:698 ast_ouraddrfor: Cannot connect [Apr 4 12:46:37] WARNING[6003]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x8379240 (len 787) to 0.0.29.200:5060 returned -1: Invalid argument -- Called 7624 -- Called 7623 -- SIP/7623-0009 is ringing [Apr 4 12:46:38] WARNING[5982]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x8379240 (len 787) to 0.0.29.200:5060 returned -1: Invalid argument [Apr 4 12:46:39] WARNING[5982]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x8379240 (len 787) to 0.0.29.200:5060 returned -1: Invalid argument [Apr 4 12:46:41] WARNING[5982]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x8379240 (len 787) to 0.0.29.200:5060 returned -1: Invalid argument [Apr 4 12:46:41] NOTICE[5975]: chan_iax2.c:4643 __auto_congest: Auto-congesting call due to slow response -- IAX2/0.0.29.199:4569-5537 is circuit-busy -- Hungup 'IAX2/0.0.29.199:4569-5537' [Apr 4 12:46:45] WARNING[5982]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x8379240 (len 787) to 0.0.29.200:5060 returned -1: Invalid argument -- SIP/7623-0009 connected line has changed. Saving it until answer
Re: [asterisk-users] WARNING chan_sip.c:3115 __sip_xmit
They are on valid IP address range and working properly but when i switched off that phone and dialing number from other phone i am getting following WARNING!! So i would like to have some thing like who check CHANNEL first and then say Phone is not register or If phone is available it will ring phone. I guess ChanIsAvail will fix my issue. http://www.asteriskguru.com/tutorials/chanisavail_image60455.jpg But now my asterisk saying i don't have cut application :( How to compile app_cut.so i didn't find anything related to this in asterisk source. -- User entered nothing. [Apr 7 16:36:53] WARNING[14134]: pbx.c:4055 pbx_extension_helper: No application 'Cut' for extension (macro-stdexten, s, 3) == Spawn extension (macro-stdexten, s, 3) exited non-zero on 'SIP/7527-003a' in macro 'stdexten' Date: Thu, 7 Apr 2011 16:40:12 -0400 From: p...@dugasenterprises.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] WARNING chan_sip.c:3115 __sip_xmit Just a guess but is it possible one of your SIP peers (7623 or 7624) has an invalid IP address of 0.0.29.200? I wonder what sip show peers shows. On Thu, Apr 7, 2011 at 4:03 PM, satish patel satish...@hotmail.com wrote: Re-opening this issue. If i dial number which doesn't existing then i am getting following error. So is there anyway i can fix my dialplan to check whether that number exist or not or i can check channel status. shirley*CLI == Using SIP RTP CoS mark 5 -- Executing [7623@from-sip:1] Macro(SIP/7527-0032, stdexten,7623,sip/7623sip/7624) in new stack -- Executing [s@macro-stdexten:1] Dial(SIP/7527-0032, sip/7623sip/7624IAX2/7623,20,t) in new stack [Apr 7 15:58:34] WARNING[14006]: app_dial.c:2039 dial_exec_full: Unable to create channel of type 'sip' (cause 20 - Unknown) == Using SIP RTP CoS mark 5 [Apr 7 15:58:34] WARNING[14006]: acl.c:698 ast_ouraddrfor: Cannot connect [Apr 7 15:58:34] WARNING[14006]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x2e08d50 (len 789) to 0.0.29.200:5060 returned -1: Invalid argument -- Called 7624 -- Called 7623 [Apr 7 15:58:34] WARNING[13920]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x2e08d50 (len 789) to 0.0.29.200:5060 returned -1: Invalid argument [Apr 7 15:58:35] WARNING[13920]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x2e08d50 (len 789) to 0.0.29.200:5060 returned -1: Invalid argument [Apr 7 15:58:37] WARNING[13920]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x2e08d50 (len 789) to 0.0.29.200:5060 returned -1: Invalid argument [Apr 7 15:58:38] NOTICE[13914]: chan_iax2.c:4643 __auto_congest: Auto-congesting call due to slow response -- IAX2/0.0.29.199:4569-13525 is circuit-busy -- Hungup 'IAX2/0.0.29.199:4569-13525' [Apr 7 15:58:39] WARNING[13920]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x7fd08e3f45a0 (len 787) to 0.0.29.200:5060 returned -1: Invalid argument [Apr 7 15:58:40] WARNING[13920]: chan_sip.c:3386 retrans_pkt: Retransmission timeout reached on transmission 6cf13d63561e7c106c31ffb74571e661@172.30.1.47:5060 for seqno 102 (Critical Request) -- See doc/sip-retransmit.txt. Packet timed out after 32000ms with no response [Apr 7 15:58:41] WARNING[13920]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x2e08d50 (len 789) to 0.0.29.200:5060 returned -1: Invalid argument == Spawn extension (macro-stdexten, s, 1) exited non-zero on 'SIP/7527-0032' in macro 'stdexten' == Spawn extension (from-sip, 7623, 1) exited non-zero on 'SIP/7527-0032' [Apr 7 15:58:49] WARNING[13920]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x2e08d50 (len 789) to 0.0.29.200:5060 returned -1: Invalid argument From: satish...@hotmail.com To: asterisk-users@lists.digium.com Date: Mon, 4 Apr 2011 20:22:55 + Subject: Re: [asterisk-users] WARNING chan_sip.c:3115 __sip_xmit Thanks for reply! I found this problem only with X-lite version of softphone. Other phones are working fine without any WARNING! look like X-lite has some short of SIP issue. -S From: mden...@gmail.com Date: Mon, 4 Apr 2011 15:59:43 -0400 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] WARNING chan_sip.c:3115 __sip_xmit On Mon, Apr 4, 2011 at 3:51 PM, satish patel satish...@hotmail.com wrote: Hey Guys, Whenever i calling any extension i am getting following WARNING messages do you have any idea they coming from where? -Satish shirley*CLI == Using SIP RTP CoS mark 5 -- Executing [7623@from-sip:1] Macro(SIP/7527-0008, stdexten,7623,sip/7623sip/7624) in new stack -- Executing [s@macro-stdexten:1] Dial(SIP/7527-0008, sip/7623sip/7624iax2/7623,20,t) in new stack == Using SIP RTP CoS mark 5 -- Called 7623 == Using SIP RTP CoS mark 5 [Apr 4 12:46:37] WARNING[6003]: acl.c:698 ast_ouraddrfor: Cannot
Re: [asterisk-users] WARNING chan_sip.c:3115 __sip_xmit
That should be CUT all caps I think -Original Message- From: satish patel satish...@hotmail.com Sender: asterisk-users-boun...@lists.digium.com Date: Thu, 7 Apr 2011 20:45:21 To: asterisk-usersasterisk-users@lists.digium.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] WARNING chan_sip.c:3115 __sip_xmit -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] WARNING chan_sip.c:3115 __sip_xmit
Yes! You are right! Its working. Now issue is we have SIP extension for local office users and same number has IAX extension for remote traveling users. How could i use ChanIsAvail with best action ? I did following exten = s,1,ChanIsAvail(${ARG2}IAX2/${ARG1},20,t) exten = s,n,NoOp(${AVAILCHAN}) exten = s,n,Set(NewVar=${CUT(AVAILCHAN,,1)}) exten = s,n,NoOp(${NewVar}) exten = s,n,Dial(${NewVar}/${EXTEN}) exten = s,n,Hangup() And in result i got following: Why its looking at IAX2/0.0.29.199 what is 0.0.29.199? shirley*CLI == Using SIP RTP CoS mark 5 -- Executing [7623@from-sip:1] Macro(SIP/7527-004c, stdexten,7623,SIP/7623) in new stack -- Executing [s@macro-stdexten:1] ChanIsAvail(SIP/7527-004c, SIP/7623IAX2/7623,20,t) in new stack -- Hungup 'IAX2/0.0.29.199:4569-2707' -- Executing [s@macro-stdexten:2] NoOp(SIP/7527-004c, IAX2/0.0.29.199:4569-2707) in new stack -- Executing [s@macro-stdexten:3] Set(SIP/7527-004c, NewVar=IAX2/0.0.29.199:4569) in new stack -- Executing [s@macro-stdexten:4] NoOp(SIP/7527-004c, IAX2/0.0.29.199:4569) in new stack -- Executing [s@macro-stdexten:5] Dial(SIP/7527-004c, IAX2/0.0.29.199:4569/s) in new stack -- Called 0.0.29.199:4569/s [Apr 7 16:59:21] NOTICE[13915]: chan_iax2.c:4643 __auto_congest: Auto-congesting call due to slow response -- IAX2/0.0.29.199:4569-3390 is circuit-busy -- Hungup 'IAX2/0.0.29.199:4569-3390' == Everyone is busy/congested at this time (1:0/1/0) -- Executing [s@macro-stdexten:6] Hangup(SIP/7527-004c, ) in new stack == Spawn extension (macro-stdexten, s, 6) exited non-zero on 'SIP/7527-004c' in macro 'stdexten' == Spawn extension (from-sip, 7623, 1) exited non-zero on 'SIP/7527-004c' To: asterisk-users@lists.digium.com From: isr...@gmail.com Date: Thu, 7 Apr 2011 20:49:04 + Subject: Re: [asterisk-users] WARNING chan_sip.c:3115 __sip_xmit That should be CUT all caps I think -Original Message- From: satish patel satish...@hotmail.com Sender: asterisk-users-boun...@lists.digium.com Date: Thu, 7 Apr 2011 20:45:21 To: asterisk-usersasterisk-users@lists.digium.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] WARNING chan_sip.c:3115 __sip_xmit -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] WARNING chan_sip.c:3115 __sip_xmit
Hey Guys, Whenever i calling any extension i am getting following WARNING messages do you have any idea they coming from where? -Satish shirley*CLI == Using SIP RTP CoS mark 5 -- Executing [7623@from-sip:1] Macro(SIP/7527-0008, stdexten,7623,sip/7623sip/7624) in new stack -- Executing [s@macro-stdexten:1] Dial(SIP/7527-0008, sip/7623sip/7624iax2/7623,20,t) in new stack == Using SIP RTP CoS mark 5 -- Called 7623 == Using SIP RTP CoS mark 5 [Apr 4 12:46:37] WARNING[6003]: acl.c:698 ast_ouraddrfor: Cannot connect [Apr 4 12:46:37] WARNING[6003]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x8379240 (len 787) to 0.0.29.200:5060 returned -1: Invalid argument -- Called 7624 -- Called 7623 -- SIP/7623-0009 is ringing [Apr 4 12:46:38] WARNING[5982]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x8379240 (len 787) to 0.0.29.200:5060 returned -1: Invalid argument [Apr 4 12:46:39] WARNING[5982]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x8379240 (len 787) to 0.0.29.200:5060 returned -1: Invalid argument [Apr 4 12:46:41] WARNING[5982]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x8379240 (len 787) to 0.0.29.200:5060 returned -1: Invalid argument [Apr 4 12:46:41] NOTICE[5975]: chan_iax2.c:4643 __auto_congest: Auto-congesting call due to slow response -- IAX2/0.0.29.199:4569-5537 is circuit-busy -- Hungup 'IAX2/0.0.29.199:4569-5537' [Apr 4 12:46:45] WARNING[5982]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x8379240 (len 787) to 0.0.29.200:5060 returned -1: Invalid argument -- SIP/7623-0009 connected line has changed. Saving it until answer for SIP/7527-0008 -- SIP/7623-0009 answered SIP/7527-0008 [Apr 4 12:46:53] WARNING[5982]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x8379240 (len 787) to 0.0.29.200:5060 returned -1: Invalid argument == Spawn extension (macro-stdexten, s, 1) exited non-zero on 'SIP/7527-0008' in macro 'stdexten' == Spawn extension (from-sip, 7623, 1) exited non-zero on 'SIP/7527-0008' [Apr 4 12:47:09] WARNING[5982]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x8379240 (len 787) to 0.0.29.200:5060 returned -1: Invalid argument [Apr 4 12:47:09] WARNING[5982]: chan_sip.c:3386 retrans_pkt: Retransmission timeout reached on transmission 23bee79c00a393995398c4d76372049e@172.30.1.47:5060 for seqno 102 (Critical Request) -- See doc/sip-retransmit.txt. Packet timed out after 32000ms with no response -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] WARNING chan_sip.c:3115 __sip_xmit
Hi Satish! Few days ago I had the same problem, and was a problem in my dialplan. Post your extensions.conf and let's see. Best regards, Fellipe From: satish...@hotmail.com To: asterisk-users@lists.digium.com Date: Mon, 4 Apr 2011 19:51:26 + Subject: [asterisk-users] WARNING chan_sip.c:3115 __sip_xmit Hey Guys, Whenever i calling any extension i am getting following WARNING messages do you have any idea they coming from where? -Satish shirley*CLI == Using SIP RTP CoS mark 5 -- Executing [7623@from-sip:1] Macro(SIP/7527-0008, stdexten,7623,sip/7623sip/7624) in new stack -- Executing [s@macro-stdexten:1] Dial(SIP/7527-0008, sip/7623sip/7624iax2/7623,20,t) in new stack == Using SIP RTP CoS mark 5 -- Called 7623 == Using SIP RTP CoS mark 5 [Apr 4 12:46:37] WARNING[6003]: acl.c:698 ast_ouraddrfor: Cannot connect [Apr 4 12:46:37] WARNING[6003]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x8379240 (len 787) to 0.0.29.200:5060 returned -1: Invalid argument -- Called 7624 -- Called 7623 -- SIP/7623-0009 is ringing [Apr 4 12:46:38] WARNING[5982]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x8379240 (len 787) to 0.0.29.200:5060 returned -1: Invalid argument [Apr 4 12:46:39] WARNING[5982]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x8379240 (len 787) to 0.0.29.200:5060 returned -1: Invalid argument [Apr 4 12:46:41] WARNING[5982]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x8379240 (len 787) to 0.0.29.200:5060 returned -1: Invalid argument [Apr 4 12:46:41] NOTICE[5975]: chan_iax2.c:4643 __auto_congest: Auto-congesting call due to slow response -- IAX2/0.0.29.199:4569-5537 is circuit-busy -- Hungup 'IAX2/0.0.29.199:4569-5537' [Apr 4 12:46:45] WARNING[5982]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x8379240 (len 787) to 0.0.29.200:5060 returned -1: Invalid argument -- SIP/7623-0009 connected line has changed. Saving it until answer for SIP/7527-0008 -- SIP/7623-0009 answered SIP/7527-0008 [Apr 4 12:46:53] WARNING[5982]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x8379240 (len 787) to 0.0.29.200:5060 returned -1: Invalid argument == Spawn extension (macro-stdexten, s, 1) exited non-zero on 'SIP/7527-0008' in macro 'stdexten' == Spawn extension (from-sip, 7623, 1) exited non-zero on 'SIP/7527-0008' [Apr 4 12:47:09] WARNING[5982]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x8379240 (len 787) to 0.0.29.200:5060 returned -1: Invalid argument [Apr 4 12:47:09] WARNING[5982]: chan_sip.c:3386 retrans_pkt: Retransmission timeout reached on transmission 23bee79c00a393995398c4d76372049e@172.30.1.47:5060 for seqno 102 (Critical Request) -- See doc/sip-retransmit.txt. Packet timed out after 32000ms with no response -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] WARNING chan_sip.c:3115 __sip_xmit
On Mon, Apr 4, 2011 at 3:51 PM, satish patel satish...@hotmail.com wrote: Hey Guys, Whenever i calling any extension i am getting following WARNING messages do you have any idea they coming from where? -Satish shirley*CLI == Using SIP RTP CoS mark 5 -- Executing [7623@from-sip:1] Macro(SIP/7527-0008, stdexten,7623,sip/7623sip/7624) in new stack -- Executing [s@macro-stdexten:1] Dial(SIP/7527-0008, sip/7623sip/7624iax2/7623,20,t) in new stack == Using SIP RTP CoS mark 5 -- Called 7623 == Using SIP RTP CoS mark 5 [Apr 4 12:46:37] WARNING[6003]: acl.c:698 ast_ouraddrfor: Cannot connect [Apr 4 12:46:37] WARNING[6003]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x8379240 (len 787) to 0.0.29.200:5060 returned -1: Invalid argument -- Called 7624 -- Called 7623 -- SIP/7623-0009 is ringing [Apr 4 12:46:38] WARNING[5982]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x8379240 (len 787) to 0.0.29.200:5060 returned -1: Invalid argument [Apr 4 12:46:39] WARNING[5982]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x8379240 (len 787) to 0.0.29.200:5060 returned -1: Invalid argument [Apr 4 12:46:41] WARNING[5982]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x8379240 (len 787) to 0.0.29.200:5060 returned -1: Invalid argument [Apr 4 12:46:41] NOTICE[5975]: chan_iax2.c:4643 __auto_congest: Auto-congesting call due to slow response -- IAX2/0.0.29.199:4569-5537 is circuit-busy -- Hungup 'IAX2/0.0.29.199:4569-5537' [Apr 4 12:46:45] WARNING[5982]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x8379240 (len 787) to 0.0.29.200:5060 returned -1: Invalid argument -- SIP/7623-0009 connected line has changed. Saving it until answer for SIP/7527-0008 -- SIP/7623-0009 answered SIP/7527-0008 [Apr 4 12:46:53] WARNING[5982]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x8379240 (len 787) to 0.0.29.200:5060 returned -1: Invalid argument == Spawn extension (macro-stdexten, s, 1) exited non-zero on 'SIP/7527-0008' in macro 'stdexten' == Spawn extension (from-sip, 7623, 1) exited non-zero on 'SIP/7527-0008' [Apr 4 12:47:09] WARNING[5982]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x8379240 (len 787) to 0.0.29.200:5060 returned -1: Invalid argument [Apr 4 12:47:09] WARNING[5982]: chan_sip.c:3386 retrans_pkt: Retransmission timeout reached on transmission 23bee79c00a393995398c4d76372049e@172.30.1.47:5060 for seqno 102 (Critical Request) -- See doc/sip-retransmit.txt. Packet timed out after 32000ms with no response Satish, Run dmesg and look for anything funny. This sounds very similar to when I had a netfilter nat helper not helping me at all. -M -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] WARNING chan_sip.c:3115 __sip_xmit
Thanks for reply! I found this problem only with X-lite version of softphone. Other phones are working fine without any WARNING! look like X-lite has some short of SIP issue. -S From: mden...@gmail.com Date: Mon, 4 Apr 2011 15:59:43 -0400 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] WARNING chan_sip.c:3115 __sip_xmit On Mon, Apr 4, 2011 at 3:51 PM, satish patel satish...@hotmail.com wrote: Hey Guys, Whenever i calling any extension i am getting following WARNING messages do you have any idea they coming from where? -Satish shirley*CLI == Using SIP RTP CoS mark 5 -- Executing [7623@from-sip:1] Macro(SIP/7527-0008, stdexten,7623,sip/7623sip/7624) in new stack -- Executing [s@macro-stdexten:1] Dial(SIP/7527-0008, sip/7623sip/7624iax2/7623,20,t) in new stack == Using SIP RTP CoS mark 5 -- Called 7623 == Using SIP RTP CoS mark 5 [Apr 4 12:46:37] WARNING[6003]: acl.c:698 ast_ouraddrfor: Cannot connect [Apr 4 12:46:37] WARNING[6003]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x8379240 (len 787) to 0.0.29.200:5060 returned -1: Invalid argument -- Called 7624 -- Called 7623 -- SIP/7623-0009 is ringing [Apr 4 12:46:38] WARNING[5982]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x8379240 (len 787) to 0.0.29.200:5060 returned -1: Invalid argument [Apr 4 12:46:39] WARNING[5982]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x8379240 (len 787) to 0.0.29.200:5060 returned -1: Invalid argument [Apr 4 12:46:41] WARNING[5982]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x8379240 (len 787) to 0.0.29.200:5060 returned -1: Invalid argument [Apr 4 12:46:41] NOTICE[5975]: chan_iax2.c:4643 __auto_congest: Auto-congesting call due to slow response -- IAX2/0.0.29.199:4569-5537 is circuit-busy -- Hungup 'IAX2/0.0.29.199:4569-5537' [Apr 4 12:46:45] WARNING[5982]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x8379240 (len 787) to 0.0.29.200:5060 returned -1: Invalid argument -- SIP/7623-0009 connected line has changed. Saving it until answer for SIP/7527-0008 -- SIP/7623-0009 answered SIP/7527-0008 [Apr 4 12:46:53] WARNING[5982]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x8379240 (len 787) to 0.0.29.200:5060 returned -1: Invalid argument == Spawn extension (macro-stdexten, s, 1) exited non-zero on 'SIP/7527-0008' in macro 'stdexten' == Spawn extension (from-sip, 7623, 1) exited non-zero on 'SIP/7527-0008' [Apr 4 12:47:09] WARNING[5982]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x8379240 (len 787) to 0.0.29.200:5060 returned -1: Invalid argument [Apr 4 12:47:09] WARNING[5982]: chan_sip.c:3386 retrans_pkt: Retransmission timeout reached on transmission 23bee79c00a393995398c4d76372049e@172.30.1.47:5060 for seqno 102 (Critical Request) -- See doc/sip-retransmit.txt. Packet timed out after 32000ms with no response Satish, Run dmesg and look for anything funny. This sounds very similar to when I had a netfilter nat helper not helping me at all. -M -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users