[Asterisk-Users] Asterisk Beta 2 Possible Bug.

[Chris Modesitt]

I am testing Asterisk Beta 2 in our lab and I have found a possible bug, the
box is setup with a T410P.  Call path looks like this:

 

T1 PRI --> Asterisk Server(1.2.0beta2) -->  SIP Interaction Proxy -->
Asterisk Server (1.0.9) --> SIP Phone.

 

This works perfectly.

 

SIP Phone --> Asterisk Server (1.0.9) --> SIP Interaction Proxy --> Asterisk
Server(1.2.0beta2) --> T1 PRI

 

No Audio either direction, there are no firewalls or nat traversals between
the any of the equipment.  If I change the Asterisk(1.2.0beta2) server back
to (1.0.9) everything works great.

 

The only error I see is generated on my Interaction SIP Proxy aka "to
retrieve next Via, don't know where to send responseSIP/2.0 200 OK" I have
included the entire message bellow.  Unfortunately I am not educated in SIP
messaging to spot the problem right off.  I would be willing to test with
anybody that would like to tackle the problem.

 

 

Chris

 

 

 

SIPSession::proxyResponseImmediately(): Failed to retrieve next Via, don't
know where to send responseSIP/2.0 200 OK

From: "Veracity Communications"
<sip:8017654321 at 192.168.201.18>;tag=as4177fb3e

To: <sip:99918011234567 at 192.168.201.10>;tag=as4a7c573e

Call-ID: 0ad778fe68521e5823395118731bb234 at 192.168.201.18

CSeq: 102 INVITE

User-Agent: Asterisk PBX

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Max-Forwards: 70

Contact: <sip:8011234567 at 192.168.201.14>

Content-Type: application/sdp

Content-Length: 214

 

v=0

o=root 1076 1077 IN IP4 192.168.201.14

s=session

c=IN IP4 192.168.201.14

t=0 0

m=audio 17268 RTP/AVP 0 101

a=rtpmap:0 PCMU/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=silenceSupp:off - - - -

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