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

satish patel satish_lx at hotmail.com
Mon Apr 18 11:45:10 CDT 2011 


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

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 ? 

12:38:46.704472 IP (tos 0x0, ttl 64, id 0, offset 0, flags [DF], proto UDP (17), length 623)
    dhcp-254-211.east.ora.com.sip > satish-desktop.sip: SIP, length: 595
        INVITE sip:7103 at 172.30.245.208:5060 SIP/2.0
        Via: SIP/2.0/UDP 172.30.254.211:5060;branch=z9hG4bK-3036-1-0
        From: sipp <sip:sipp at 172.30.254.211:5060>;tag=3036SIPpTag091
        To: sut <sip:7103 at 172.30.245.208:5060>
        Call-ID: 1-3036 at 172.30.254.211
        CSeq: 1 INVITE
        Contact: sip:sipp at 172.30.254.211:5060
        Max-Forwards: 70
        Subject: Performance Test
        Content-Type: application/sdp
        Content-Length:   198

        v=0
        o=user1 53655765 2353687637 IN IP4 172.30.254.211
        s=-
        c=IN IP4 172.30.254.211
        t=0 0
        m=audio 6000 RTP/AVP 8 101
        a=rtpmap:8 PCMA/8000
        a=rtpmap:101 telephone-event/8000
        a=fmtp:101 0-11,16

12:38:46.705445 IP (tos 0x0, ttl 64, id 1416, offset 0, flags [none], proto UDP (17), length 487)
    satish-desktop.sip > dhcp-254-211.east.ora.com.sip: SIP, length: 459
        SIP/2.0 100 Trying
        Via: SIP/2.0/UDP 172.30.254.211:5060;branch=z9hG4bK-3036-1-0;received=172.30.254.211
        From: sipp <sip:sipp at 172.30.254.211:5060>;tag=3036SIPpTag091
        To: sut <sip:7103 at 172.30.245.208:5060>
        Call-ID: 1-3036 at 172.30.254.211
        CSeq: 1 INVITE
        Server: Asterisk PBX 1.8.3.2
        Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
        Supported: replaces, timer
        Contact: <sip:7103 at 172.30.245.208:5060>
        Content-Length: 0


12:39:18.706783 IP (tos 0x0, ttl 64, id 1417, offset 0, flags [none], proto UDP (17), length 771)
    satish-desktop.sip > dhcp-254-211.east.ora.com.sip: SIP, length: 743
        SIP/2.0 200 OK
        Via: SIP/2.0/UDP 172.30.254.211:5060;branch=z9hG4bK-3036-1-0;received=172.30.254.211
        From: sipp <sip:sipp at 172.30.254.211:5060>;tag=3036SIPpTag091
        To: sut <sip:7103 at 172.30.245.208:5060>;tag=as7403b6f3
        Call-ID: 1-3036 at 172.30.254.211
        CSeq: 1 INVITE
        Server: Asterisk PBX 1.8.3.2
        Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
        Supported: replaces, timer
        Contact: <sip:7103 at 172.30.245.208:5060>
        Content-Type: application/sdp
        Content-Length: 240

        v=0
        o=root 1076282210 1076282210 IN IP4 172.30.245.208
        s=Asterisk PBX 1.8.3.2
        c=IN IP4 172.30.245.208
        t=0 0
        m=audio 17450 RTP/AVP 8 101
        a=rtpmap:8 PCMA/8000
        a=rtpmap:101 telephone-event/8000
        a=fmtp:101 0-16
        a=ptime:20
        a=sendrecv




> Date: Thu, 7 Apr 2011 16:40:12 -0400
> From: paul at dugasenterprises.com
> To: asterisk-users at 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_lx at 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 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
> >>
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> >> http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> > -- _____________________________________________________________________ --
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> 
> --
> _____________________________________________________________________
> -- 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
> 
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