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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. <br><br>I guess <span style="font-family: monospace;">ChanIsAvail will fix my issue. http://www.asteriskguru.com/tutorials/chanisavail_image60455.jpg<br><br>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.<br><br> -- User entered nothing.<br>[Apr 7 16:36:53] WARNING[14134]: pbx.c:4055 pbx_extension_helper: No application 'Cut' for extension (macro-stdexten, s, 3)<br> == Spawn extension (macro-stdexten, s, 3) exited non-zero on 'SIP/7527-0000003a' in macro 'stdexten'<br><br><br><br></span>
<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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