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:[EMAIL PROTECTED];user=phone;phone-context=unknown SIP/2.0
Via: SIP/2.0/UDP  213.160.252.50:53893
From: "611012210" <sip:[EMAIL PROTECTED]>
To: <sip:[EMAIL PROTECTED];user=phone;phone-context=unknown>
Date: Wed, 30 Jul 2003 09:26:11 GMT
Call-ID: [EMAIL PROTECTED]
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:[EMAIL PROTECTED]: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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