Look at this sip debug its saying something related Retransmitting #1 (no NAT)
to 0.0.29.200:5060:
<------------>
-- Executing [7624@from-sip:1] Macro("SIP/7527-000000c2",
"stdexten,7624,SIP/7624") in new stack
-- Executing [s@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:[email protected]>;tag=as6f6822ba
To: <sip:7624>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]: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:[email protected]>;tag=as6f6822ba
To: <sip:7624>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]: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: [email protected]
> To: [email protected]
> 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@from-sip:1] Macro("SIP/7527-0000006b",
> > "stdexten,7623,SIP/7623") in new stack
> > -- Executing [s@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@macro-stdexten:2] NoOp("SIP/7527-0000006b",
> > "IAX2/0.0.29.199:4569-5255") in new stack
> > -- Executing [s@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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