[asterisk-users] IAX2/0.0.29.199

[satish patel]

Look at this sip debug its saying something related Retransmitting #1 (no NAT) to 0.0.29.200:5060:

<------------>
    -- Executing [7624 at from-sip:1] Macro("SIP/7527-000000c2", "stdexten,7624,SIP/7624") in new stack
    -- Executing [s at macro-stdexten:1] Dial("SIP/7527-000000c2", "SIP/7624&IAX2/7624,20,t") in new stack
  == Using SIP RTP CoS mark 5
[Apr  8 12:20:53] WARNING[15194]: acl.c:698 ast_ouraddrfor: Cannot connect
Audio is at 5060
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 0.0.29.200:5060:
INVITE sip:7624 SIP/2.0
Via: SIP/2.0/UDP 172.30.1.47:5060;branch=z9hG4bK3af914e2
Max-Forwards: 70
From: "Cambridge Guest" <sip:7527 at 172.30.1.47>;tag=as6f6822ba
To: <sip:7624>
Contact: <sip:7527 at 172.30.1.47:5060>
Call-ID: 0ca7784d38d29be168f8f85711c43e4f at 172.30.1.47:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.3.2
Date: Fri, 08 Apr 2011 19:20:53 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 257

v=0
o=root 1407056235 1407056235 IN IP4 172.30.1.47
s=Asterisk PBX 1.8.3.2
c=IN IP4 172.30.1.47
t=0 0
m=audio 16720 RTP/AVP 0 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
[Apr  8 12:20:53] WARNING[15194]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x2ef3f00 (len 789) to 0.0.29.200:5060 returned -1: Invalid argument
    -- Called 7624
    -- Called 7624
Retransmitting #1 (no NAT) to 0.0.29.200:5060:
INVITE sip:7624 SIP/2.0
Via: SIP/2.0/UDP 172.30.1.47:5060;branch=z9hG4bK3af914e2
Max-Forwards: 70
From: "Cambridge Guest" <sip:7527 at 172.30.1.47>;tag=as6f6822ba
To: <sip:7624>
Contact: <sip:7527 at 172.30.1.47:5060>
Call-ID: 0ca7784d38d29be168f8f85711c43e4f at 172.30.1.47:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.3.2
Date: Fri, 08 Apr 2011 19:20:53 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 257




> Date: Fri, 8 Apr 2011 11:12:59 -0400
> From: pabelanger at digium.com
> To: asterisk-users at lists.digium.com
> Subject: Re: [asterisk-users] IAX2/0.0.29.199
> 
> On 11-04-08 10:48 AM, satish patel wrote:
> >
> > Where this revers IP comes from ?
> >
> >    == Using SIP RTP CoS mark 5
> >      -- Executing [7623 at from-sip:1] Macro("SIP/7527-0000006b", "stdexten,7623,SIP/7623") in new stack
> >      -- Executing [s at macro-stdexten:1] ChanIsAvail("SIP/7527-0000006b", "SIP/7623&IAX2/7623,20,t") in new stack
> >      -- Hungup 'IAX2/0.0.29.199:4569-5255'
> >      -- Executing [s at macro-stdexten:2] NoOp("SIP/7527-0000006b", "IAX2/0.0.29.199:4569-5255") in new stack
> >      -- Executing [s at macro-stdexten:3] NoOp("SIP/7527-0000006b", "0&0") in new stack
> >      -- Auto fallthrough, channel 'SIP/7527-0000006b' status is 'UNKNOWN'
> >
> Asterisk 1.8?  Are you using realtime?  Looks to be an issue with 
> netsock2.c.
> 
> -- 
> Paul Belanger
> Digium, Inc. | Software Developer
> twitter: pabelanger | IRC: pabelanger (Freenode)
> Check us out at: http://digium.com & http://asterisk.org
> 
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