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/7527-0000003a' 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/7527-00000032", > > "stdexten,7623,sip/7623&sip/7624") in new stack > > -- Executing [s@macro-stdexten:1] Dial("SIP/7527-00000032", > > "sip/7623&sip/7624&IAX2/7623,20,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/7527-00000032' in macro 'stdexten' > > == Spawn extension (from-sip, 7623, 1) exited non-zero on > > 'SIP/7527-00000032' > > [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 +0000 > > 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/7527-00000008", > >> > "stdexten,7623,sip/7623&sip/7624") in new stack > >> > -- Executing [s@macro-stdexten:1] Dial("SIP/7527-00000008", > >> > "sip/7623&sip/7624&iax2/7623,20,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/7623-00000009 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/7623-00000009 connected line has changed. Saving it until > >> > answer > >> > for SIP/7527-00000008 > >> > -- SIP/7623-00000009 answered SIP/7527-00000008 > >> > [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/7527-00000008' in macro 'stdexten' > >> > == Spawn extension (from-sip, 7623, 1) exited non-zero on > >> > 'SIP/7527-00000008' > >> > [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 > > -- > > _____________________________________________________________________ > > -- 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