[Asterisk-Users] chan_sip.c problems problems from cvs 1.134
Brenton D. Rothchild
brothchild at dstorage.com
Wed Jul 30 07:15:12 MST 2003
That also worked for me. My AudioCodes MP-104 FXO has no problem
making inbound calls now.
Thanks Patrick and Adam.
-Brenton
----- Original Message -----
From: "Low, Adam" <ALow at Prioritytelecom.com>
To: <asterisk-users at lists.digium.com>
Sent: Wednesday, July 30, 2003 8:45 AM
Subject: RE: [Asterisk-Users] chan_sip.c problems problems from cvs 1.134
> Well found Patrick, that did the trick for me as well !
>
> I had been trying to debug 1.135 where this portion of code wasn't added
yet ... thats a lesson learnt ...
>
> -----Original Message-----
> From: Patrick
> To: 'asterisk-users at lists.digium.com '
> Sent: 30/07/03 15:04
> Subject: RE: [Asterisk-Users] chan_sip.c problems problems from cvs 1.134
>
>
> 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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