Hi,
I am using the latest cvs release of asterisk, and the behaviour is in fact
the same,
outbound calls work fine,
but for inbound calls (from C2651 over PSTN) , SIP messages get "blocked"
by asterisk, and never reach the phone.
The setup is the same : 7960 <------> asterisk <------> C2651<-----> PSTN
Yves
|---------+------------------------------------->
| | "Low, Adam" |
| | <[EMAIL PROTECTED]>|
| | Sent by: |
| | [EMAIL PROTECTED]|
| | .digium.com |
| | |
| | |
| | 30/07/2003 11:37 |
| | Please respond to |
| | asterisk-users |
| | |
|---------+------------------------------------->
>-----------------------------------------------------------------------------------------------------------------------|
|
|
| To: "'[EMAIL PROTECTED]'" <[EMAIL PROTECTED]>
|
| cc:
|
| Subject: [Asterisk-Users] chan_sip.c problems problems from cvs 1.134
|
>-----------------------------------------------------------------------------------------------------------------------|
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
********* DISCLAIMER *********
This message and any attachment are confidential and may be privileged or
otherwise protected from disclosure and may include proprietary
information. If you are not the intended recipient, please telephone or
email the sender and delete this message and any attachment from your
system. If you are not the intended recipient you must not copy this
message or attachment or disclose the contents to any other person
_______________________________________________
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
_______________________________________________
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
