[Asterisk-Users] chan_sip.c problems problems from cvs 1.134

[James Sizemore]

I also had the same problem with sip,  I also moved back a couple of 
weeks in cvs.
I also use a AS5300 Cisco in my call chain.

I got a bunch of "Ignoring this request" in debug. I have not had time
to trace the call path on this problem yet.
 

Low, Adam wrote:

>All,
>
>I've found problems in my setup with the latest couple of revisions (1.135/1.136) of asterisk/channels/chan_sip.c In my setup I have a RH9 asterisk server, AS5300 (single E1 to PSTN) and a dozen 7940's, everything is in the same VLAN and only running SIP.
>
>Outbound calls work fine: 7940 -SIP-> Asterisk -SIP-> AS5300
>
>But inbound calls fail, I see the initial INVITE from the AS5300 which is received by asterisk but not responded to and then the AS5300 sends another few INVITE's which are received but ignored assumable as they were duplicates for the first.
>
>Unfortunately since I've been trying the different cvs revisions of chan_sip.c I've got susbequent problems with the server crashing after the first INVITE from the AS5300 using anything greater than cvs 1.134
>
>I suspect this is something to do with the per-user limits added in cvs 1.135 but I am curious to see if anyone has any problems with the latest cvs elease of asterisk with SIP ?
>
>Adam
>
>Sip read: 
>INVITE sip:4842 at 213.160.252.2;user=phone;phone-context=unknown SIP/2.0
>Via: SIP/2.0/UDP  213.160.252.50:53893
>From: "611012210" <sip:611012210 at 213.160.252.50>
>To: <sip:4842 at 213.160.252.2;user=phone;phone-context=unknown>
>Date: Wed, 30 Jul 2003 09:26:11 GMT
>Call-ID: 635D27D4-CB1D0233-0-8E9DB84 at 213.160.252.50
>Cisco-Guid: 1667049428-3407675953-0-149543808
>User-Agent: Cisco VoIP Gateway/ IOS 12.x/ SIP enabled
>CSeq: 101 INVITE
>Max-Forwards: 6
>Timestamp: 1059557171
>Contact: <sip:611012210 at 213.160.252.50:5060;user=phone>
>Expires: 180
>Content-Type: application/sdp
>Content-Length: 149
>
>v=0
>o=CiscoSystemsSIP-GW-UserAgent 5241 1607 IN IP4 213.160.252.50
>s=SIP Call
>c=IN IP4 213.160.252.50
>t=0 0
>m=audio 20032 RTP/AVP 8 0 65535 18
>
>15 headers, 6 lines
>Using latest request as basis request
>Sending to 213.160.252.50 : 53893 (non-NAT)
>Found audio format 8
>Found audio format 0
>Found audio format 65535
>Found audio format 18
>Capabilities: us - 524302, them - 268/0, combined - 12
>Non-codec capabilities: us - 1, them - 0, combined - 0
>AM00CM01*CLI> 
>Disconnected from Asterisk server
>
>
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