[Asterisk-Users] chan_sip.c problems problems from cvs 1.134
Patrick
patrick at sip2.dmv.com
Wed Jul 30 06:04:31 MST 2003
It is in the find_user() routine. If it is not an extension on the PBX,
it should return a zero
if ( isfound ) {
ast_log(LOG_DEBUG, "%s is not a local user\n", name);
ast_pthread_mutex_unlock(&userl.lock);
return 1; <--- this is the problem - change it to a 0.
}
It isn't an error, so it should just return. Change that and the function
will work properly. I tested it using an AS5350 and successly made an
inbound call.
Patrick
On Wed, 30 Jul 2003, Low, Adam wrote:
> Brenton, Yves, ...
>
> I've located the cause of the problem in chan_sip.c but am still trying to find the exact cause being completely new to the asterisk code. It seems that there was an added function in 1.135 called 'find_user' that is supposed to lookup the users incoming call limit but the routine is unable to find a matching user for my AS5300 which I suspect is because it does not REGISTER with the server prior to attempting to send calls.
>
> I'm going to continue debugging a little later and see if I can narrow it down more ...
>
> Adam
>
> -----Original Message-----
> From: yves.schaaf at restena.lu
> To: asterisk-users at lists.digium.com
> Sent: 30/07/03 14:09
> Subject: Re: [Asterisk-Users] chan_sip.c problems problems from cvs 1.134
>
>
> 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" |
> | | <ALow at Prioritytelecom.com>|
> | | Sent by: |
> | | asterisk-users-admin at lists|
> | | .digium.com |
> | | |
> | | |
> | | 30/07/2003 11:37 |
> | | Please respond to |
> | | asterisk-users |
> | | |
> |---------+------------------------------------->
>
> >-----------------------------------------------------------------------
> ------------------------------------------------|
> |
> |
> | To: "'asterisk-users at lists.digium.com'"
> <asterisk-users at lists.digium.com> |
> | 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: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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