[Asterisk-Users] sip channel driver causes asterisk to crash when talking to quintum A800

[Stephen Davies]

On Sat, 7 Jun 2003, shido wrote:

> This is the sip debug when the call went through........
> 
> Sip read:
> INVITE sip:2877433 at 64.42.218.157 SIP/2.0
> Call-ID: call-80B9AC74-7A97-D711-0006 at 64.42.218.146
> Contact: <sip:2044808000 at 64.42.218.146>
> Content-Length: 157
> Content-Type: application/sdp
> CSeq: 1 INVITE
> From: <sip:2044808000 at 64.42.218.146>;tag=402ada92-5
> To: <sip:2877433 at 64.42.218.157>
> User-Agent: Quintum/1.0.0
> Via: SIP/2.0/UDP 64.42.218.146;branch=z9hG4bK-tenor-64.42.218.146-5
> Quintum: 0c01030b0239380501
> 
> v=0
> o=Quintum 4 4 IN IP4 64.42.218.146
> s=VoipCall
> c=IN IP4 64.42.218.146
> t=0 0
> m=audio 10240 RTP/AVP 0
> c=IN IP4 64.42.218.146
> a=rtpmap:0 pcmu/8000/1
> 
> 11 headers, 8 lines
> Using latest request as basis request
> Sending to 64.42.218.146 : 5060 (non-NAT)
> Capabilities: us - 4, them - 4, combined - 4
> Non-codec capabilities: us - 1, them - 0, combined - 0
> 
> 


Funnily enough I've been looking at the same problem.  Will get a
chance to look a bit more tomorrow.

Steve