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<br>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<br><br>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 ? <br><br>12:38:46.704472 IP (tos 0x0, ttl 64, id 0, offset 0, flags [DF], proto UDP (17), length 623)<br> dhcp-254-211.east.ora.com.sip > satish-desktop.sip: SIP, length: 595<br> INVITE sip:7103@172.30.245.208:5060 SIP/2.0<br> Via: SIP/2.0/UDP 172.30.254.211:5060;branch=z9hG4bK-3036-1-0<br> From: sipp <sip:sipp@172.30.254.211:5060>;tag=3036SIPpTag091<br> To: sut <sip:7103@172.30.245.208:5060><br> Call-ID: 1-3036@172.30.254.211<br> CSeq: 1 INVITE<br> Contact: sip:sipp@172.30.254.211:5060<br> Max-Forwards: 70<br> Subject: Performance Test<br> Content-Type: application/sdp<br> Content-Length: 198<br><br> v=0<br> o=user1 53655765 2353687637 IN IP4 172.30.254.211<br> s=-<br> c=IN IP4 172.30.254.211<br> t=0 0<br> m=audio 6000 RTP/AVP 8 101<br> a=rtpmap:8 PCMA/8000<br> a=rtpmap:101 telephone-event/8000<br> a=fmtp:101 0-11,16<br><br>12:38:46.705445 IP (tos 0x0, ttl 64, id 1416, offset 0, flags [none], proto UDP (17), length 487)<br> satish-desktop.sip > dhcp-254-211.east.ora.com.sip: SIP, length: 459<br> SIP/2.0 100 Trying<br> Via: SIP/2.0/UDP 172.30.254.211:5060;branch=z9hG4bK-3036-1-0;received=172.30.254.211<br> From: sipp <sip:sipp@172.30.254.211:5060>;tag=3036SIPpTag091<br> To: sut <sip:7103@172.30.245.208:5060><br> Call-ID: 1-3036@172.30.254.211<br> CSeq: 1 INVITE<br> Server: Asterisk PBX 1.8.3.2<br> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH<br> Supported: replaces, timer<br> Contact: <sip:7103@172.30.245.208:5060><br> Content-Length: 0<br><br><br>12:39:18.706783 IP (tos 0x0, ttl 64, id 1417, offset 0, flags [none], proto UDP (17), length 771)<br> satish-desktop.sip > dhcp-254-211.east.ora.com.sip: SIP, length: 743<br> SIP/2.0 200 OK<br> Via: SIP/2.0/UDP 172.30.254.211:5060;branch=z9hG4bK-3036-1-0;received=172.30.254.211<br> From: sipp <sip:sipp@172.30.254.211:5060>;tag=3036SIPpTag091<br> To: sut <sip:7103@172.30.245.208:5060>;tag=as7403b6f3<br> Call-ID: 1-3036@172.30.254.211<br> CSeq: 1 INVITE<br> Server: Asterisk PBX 1.8.3.2<br> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH<br> Supported: replaces, timer<br> Contact: <sip:7103@172.30.245.208:5060><br> Content-Type: application/sdp<br> Content-Length: 240<br><br> v=0<br> o=root 1076282210 1076282210 IN IP4 172.30.245.208<br> s=Asterisk PBX 1.8.3.2<br> c=IN IP4 172.30.245.208<br> t=0 0<br> m=audio 17450 RTP/AVP 8 101<br> a=rtpmap:8 PCMA/8000<br> a=rtpmap:101 telephone-event/8000<br> a=fmtp:101 0-16<br> a=ptime:20<br> a=sendrecv<br><br><br><br><br>> Date: Thu, 7 Apr 2011 16:40:12 -0400<br>> From: paul@dugasenterprises.com<br>> To: asterisk-users@lists.digium.com<br>> Subject: Re: [asterisk-users] WARNING chan_sip.c:3115 __sip_xmit<br>> <br>> Just a guess but is it possible one of your SIP peers (7623 or 7624)<br>> has an invalid IP address of 0.0.29.200? I wonder what "sip show<br>> peers" shows.<br>> <br>> <br>> On Thu, Apr 7, 2011 at 4:03 PM, satish patel <satish_lx@hotmail.com> wrote:<br>> ><br>> > Re-opening this issue.<br>> ><br>> > If i dial number which doesn't existing then i am getting following error.<br>> > So is there anyway i can fix my dialplan to check whether that number exist<br>> > or not or i can check channel status.<br>> ><br>> ><br>> ><br>> > shirley*CLI><br>> > == Using SIP RTP CoS mark 5<br>> > -- Executing [7623@from-sip:1] Macro("SIP/7527-00000032",<br>> > "stdexten,7623,sip/7623&sip/7624") in new stack<br>> > -- Executing [s@macro-stdexten:1] Dial("SIP/7527-00000032",<br>> > "sip/7623&sip/7624&IAX2/7623,20,t") in new stack<br>> > [Apr 7 15:58:34] WARNING[14006]: app_dial.c:2039 dial_exec_full: Unable to<br>> > create channel of type 'sip' (cause 20 - Unknown)<br>> > == Using SIP RTP CoS mark 5<br>> > [Apr 7 15:58:34] WARNING[14006]: acl.c:698 ast_ouraddrfor: Cannot connect<br>> > [Apr 7 15:58:34] WARNING[14006]: chan_sip.c:3115 __sip_xmit: sip_xmit of<br>> > 0x2e08d50 (len 789) to 0.0.29.200:5060 returned -1: Invalid argument<br>> > -- Called 7624<br>> > -- Called 7623<br>> > [Apr 7 15:58:34] WARNING[13920]: chan_sip.c:3115 __sip_xmit: sip_xmit of<br>> > 0x2e08d50 (len 789) to 0.0.29.200:5060 returned -1: Invalid argument<br>> > [Apr 7 15:58:35] WARNING[13920]: chan_sip.c:3115 __sip_xmit: sip_xmit of<br>> > 0x2e08d50 (len 789) to 0.0.29.200:5060 returned -1: Invalid argument<br>> > [Apr 7 15:58:37] WARNING[13920]: chan_sip.c:3115 __sip_xmit: sip_xmit of<br>> > 0x2e08d50 (len 789) to 0.0.29.200:5060 returned -1: Invalid argument<br>> > [Apr 7 15:58:38] NOTICE[13914]: chan_iax2.c:4643 __auto_congest:<br>> > Auto-congesting call due to slow response<br>> > -- IAX2/0.0.29.199:4569-13525 is circuit-busy<br>> > -- Hungup 'IAX2/0.0.29.199:4569-13525'<br>> > [Apr 7 15:58:39] WARNING[13920]: chan_sip.c:3115 __sip_xmit: sip_xmit of<br>> > 0x7fd08e3f45a0 (len 787) to 0.0.29.200:5060 returned -1: Invalid argument<br>> > [Apr 7 15:58:40] WARNING[13920]: chan_sip.c:3386 retrans_pkt:<br>> > Retransmission timeout reached on transmission<br>> > 6cf13d63561e7c106c31ffb74571e661@172.30.1.47:5060 for seqno 102 (Critical<br>> > Request) -- See doc/sip-retransmit.txt.<br>> > Packet timed out after 32000ms with no response<br>> > [Apr 7 15:58:41] WARNING[13920]: chan_sip.c:3115 __sip_xmit: sip_xmit of<br>> > 0x2e08d50 (len 789) to 0.0.29.200:5060 returned -1: Invalid argument<br>> > == Spawn extension (macro-stdexten, s, 1) exited non-zero on<br>> > 'SIP/7527-00000032' in macro 'stdexten'<br>> > == Spawn extension (from-sip, 7623, 1) exited non-zero on<br>> > 'SIP/7527-00000032'<br>> > [Apr 7 15:58:49] WARNING[13920]: chan_sip.c:3115 __sip_xmit: sip_xmit of<br>> > 0x2e08d50 (len 789) to 0.0.29.200:5060 returned -1: Invalid argument<br>> ><br>> ><br>> ><br>> ><br>> > ________________________________<br>> > From: satish_lx@hotmail.com<br>> > To: asterisk-users@lists.digium.com<br>> > Date: Mon, 4 Apr 2011 20:22:55 +0000<br>> > Subject: Re: [asterisk-users] WARNING chan_sip.c:3115 __sip_xmit<br>> ><br>> ><br>> > Thanks for reply!<br>> ><br>> > I found this problem only with X-lite version of softphone. Other phones<br>> > are working fine without any WARNING! look like X-lite has some short of<br>> > SIP issue.<br>> ><br>> > -S<br>> ><br>> ><br>> ><br>> >> From: mdeneen@gmail.com<br>> >> Date: Mon, 4 Apr 2011 15:59:43 -0400<br>> >> To: asterisk-users@lists.digium.com<br>> >> Subject: Re: [asterisk-users] WARNING chan_sip.c:3115 __sip_xmit<br>> >><br>> >> On Mon, Apr 4, 2011 at 3:51 PM, satish patel <satish_lx@hotmail.com><br>> >> wrote:<br>> >> ><br>> >> > Hey Guys,<br>> >> ><br>> >> > Whenever i calling any extension i am getting following WARNING messages<br>> >> > do<br>> >> > you have any idea they coming from where?<br>> >> ><br>> >> > -Satish<br>> >> ><br>> >> ><br>> >> ><br>> >> > shirley*CLI><br>> >> > == Using SIP RTP CoS mark 5<br>> >> > -- Executing [7623@from-sip:1] Macro("SIP/7527-00000008",<br>> >> > "stdexten,7623,sip/7623&sip/7624") in new stack<br>> >> > -- Executing [s@macro-stdexten:1] Dial("SIP/7527-00000008",<br>> >> > "sip/7623&sip/7624&iax2/7623,20,t") in new stack<br>> >> > == Using SIP RTP CoS mark 5<br>> >> > -- Called 7623<br>> >> > == Using SIP RTP CoS mark 5<br>> >> > [Apr 4 12:46:37] WARNING[6003]: acl.c:698 ast_ouraddrfor: Cannot<br>> >> > connect<br>> >> > [Apr 4 12:46:37] WARNING[6003]: chan_sip.c:3115 __sip_xmit: sip_xmit of<br>> >> > 0x8379240 (len 787) to 0.0.29.200:5060 returned -1: Invalid argument<br>> >> > -- Called 7624<br>> >> > -- Called 7623<br>> >> > -- SIP/7623-00000009 is ringing<br>> >> > [Apr 4 12:46:38] WARNING[5982]: chan_sip.c:3115 __sip_xmit: sip_xmit of<br>> >> > 0x8379240 (len 787) to 0.0.29.200:5060 returned -1: Invalid argument<br>> >> > [Apr 4 12:46:39] WARNING[5982]: chan_sip.c:3115 __sip_xmit: sip_xmit of<br>> >> > 0x8379240 (len 787) to 0.0.29.200:5060 returned -1: Invalid argument<br>> >> > [Apr 4 12:46:41] WARNING[5982]: chan_sip.c:3115 __sip_xmit: sip_xmit of<br>> >> > 0x8379240 (len 787) to 0.0.29.200:5060 returned -1: Invalid argument<br>> >> > [Apr 4 12:46:41] NOTICE[5975]: chan_iax2.c:4643 __auto_congest:<br>> >> > Auto-congesting call due to slow response<br>> >> > -- IAX2/0.0.29.199:4569-5537 is circuit-busy<br>> >> > -- Hungup 'IAX2/0.0.29.199:4569-5537'<br>> >> > [Apr 4 12:46:45] WARNING[5982]: chan_sip.c:3115 __sip_xmit: sip_xmit of<br>> >> > 0x8379240 (len 787) to 0.0.29.200:5060 returned -1: Invalid argument<br>> >> > -- SIP/7623-00000009 connected line has changed. Saving it until<br>> >> > answer<br>> >> > for SIP/7527-00000008<br>> >> > -- SIP/7623-00000009 answered SIP/7527-00000008<br>> >> > [Apr 4 12:46:53] WARNING[5982]: chan_sip.c:3115 __sip_xmit: sip_xmit of<br>> >> > 0x8379240 (len 787) to 0.0.29.200:5060 returned -1: Invalid argument<br>> >> > == Spawn extension (macro-stdexten, s, 1) exited non-zero on<br>> >> > 'SIP/7527-00000008' in macro 'stdexten'<br>> >> > == Spawn extension (from-sip, 7623, 1) exited non-zero on<br>> >> > 'SIP/7527-00000008'<br>> >> > [Apr 4 12:47:09] WARNING[5982]: chan_sip.c:3115 __sip_xmit: sip_xmit of<br>> >> > 0x8379240 (len 787) to 0.0.29.200:5060 returned -1: Invalid argument<br>> >> > [Apr 4 12:47:09] WARNING[5982]: chan_sip.c:3386 retrans_pkt:<br>> >> > Retransmission<br>> >> > timeout reached on transmission<br>> >> > 23bee79c00a393995398c4d76372049e@172.30.1.47:5060 for seqno 102<br>> >> > (Critical<br>> >> > Request) -- See doc/sip-retransmit.txt.<br>> >> > Packet timed out after 32000ms with no response<br>> >> ><br>> >> ><br>> >><br>> >> Satish,<br>> >><br>> >> Run dmesg and look for anything funny. This sounds very similar to<br>> >> when I had a netfilter nat "helper" not helping me at all.<br>> >><br>> >> -M<br>> >><br>> >> --<br>> >> _____________________________________________________________________<br>> >> -- Bandwidth and Colocation Provided by http://www.api-digital.com --<br>> >> New to Asterisk? Join us for a live introductory webinar every Thurs:<br>> >> http://www.asterisk.org/hello<br>> >><br>> >> asterisk-users mailing list<br>> >> To UNSUBSCRIBE or update options visit:<br>> >> http://lists.digium.com/mailman/listinfo/asterisk-users<br>> ><br>> > -- _____________________________________________________________________ --<br>> > Bandwidth and Colocation Provided by http://www.api-digital.com -- New to<br>> > Asterisk? Join us for a live introductory webinar every Thurs:<br>> > http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or<br>> > update options visit:<br>> > http://lists.digium.com/mailman/listinfo/asterisk-users<br>> > --<br>> > _____________________________________________________________________<br>> > -- Bandwidth and Colocation Provided by http://www.api-digital.com --<br>> > New to Asterisk? Join us for a live introductory webinar every Thurs:<br>> > http://www.asterisk.org/hello<br>> ><br>> > asterisk-users mailing list<br>> > To UNSUBSCRIBE or update options visit:<br>> > http://lists.digium.com/mailman/listinfo/asterisk-users<br>> ><br>> <br>> --<br>> _____________________________________________________________________<br>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --<br>> New to Asterisk? Join us for a live introductory webinar every Thurs:<br>> http://www.asterisk.org/hello<br>> <br>> asterisk-users mailing list<br>> To UNSUBSCRIBE or update options visit:<br>> http://lists.digium.com/mailman/listinfo/asterisk-users<br> </body>
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