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<br>Re-opening this issue. <br><br>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.<br><br><br><br>shirley*CLI><br> == Using SIP RTP CoS mark 5<br> -- Executing [7623@from-sip:1] Macro("SIP/7527-00000032", "stdexten,7623,sip/7623&sip/7624") in new stack<br> -- Executing [s@macro-stdexten:1] Dial("SIP/7527-00000032", "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 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 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 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 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 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: 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 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: Retransmission timeout reached on transmission 6cf13d63561e7c106c31ffb74571e661@172.30.1.47:5060 for seqno 102 (Critical 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 0x2e08d50 (len 789) to 0.0.29.200:5060 returned -1: Invalid argument<br> == Spawn extension (macro-stdexten, s, 1) exited non-zero on 'SIP/7527-00000032' in macro 'stdexten'<br> == Spawn extension (from-sip, 7623, 1) exited non-zero on 'SIP/7527-00000032'<br>[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<br><br><br><br><br><hr id="stopSpelling">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>
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<br>Thanks for reply! <br><br>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. <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> wrote:<br>> ><br>> > Hey Guys,<br>> ><br>> > Whenever i calling any extension i am getting following WARNING messages 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 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 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: Retransmission<br>> > timeout reached on transmission<br>> > 23bee79c00a393995398c4d76372049e@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>> ><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>
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