[asterisk-users] WARNING chan_sip.c:3115 __sip_xmit

satish patel satish_lx at hotmail.com
Thu Apr 7 15:03:28 CDT 2011



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 at from-sip:1] Macro("SIP/7527-00000032", "stdexten,7623,sip/7623&sip/7624") in new stack
    -- Executing [s at 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 at 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_lx at hotmail.com
To: asterisk-users at 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: mdeneen at gmail.com
> Date: Mon, 4 Apr 2011 15:59:43 -0400
> To: asterisk-users at 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_lx at 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 at from-sip:1] Macro("SIP/7527-00000008",
> > "stdexten,7623,sip/7623&sip/7624") in new stack
> >     -- Executing [s at 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 at 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 		 	   		  
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110407/cc58b07f/attachment.htm>


More information about the asterisk-users mailing list