[Asterisk-Users] chan_sip.c problems problems from cvs 1.134

yves.schaaf at restena.lu yves.schaaf at restena.lu
Wed Jul 30 22:56:00 MST 2003


Inbound calls work pretty fine again,
Thanks for you help

Yves



|---------+------------------------------------->
|         |           "Brenton D. Rothchild"    |
|         |           <brothchild at dstorage.com> |
|         |           Sent by:                  |
|         |           asterisk-users-admin at lists|
|         |           .digium.com               |
|         |                                     |
|         |                                     |
|         |           30/07/2003 16:15          |
|         |           Please respond to         |
|         |           asterisk-users            |
|         |                                     |
|---------+------------------------------------->
  >-----------------------------------------------------------------------------------------------------------------------|
  |                                                                                                                       |
  |       To:       <asterisk-users at lists.digium.com>                                                                     |
  |       cc:                                                                                                             |
  |       Subject:  Re: [Asterisk-Users] chan_sip.c problems problems from cvs 1.134                                      |
  >-----------------------------------------------------------------------------------------------------------------------|




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
> >
> >
> > ********* 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
> > Asterisk-Users at lists.digium.com
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> >
> >
> >
> >
> > _______________________________________________
> > Asterisk-Users mailing list
> > Asterisk-Users at lists.digium.com
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> >
> > ********* 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
> > Asterisk-Users at lists.digium.com
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
>
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
> ********* 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
> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users


_______________________________________________
Asterisk-Users mailing list
Asterisk-Users at lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users








More information about the asterisk-users mailing list