Hi all,
I am using the openserctl command to add users to the Openser database and I can verify if these numbers are added to the Subscriber table in the mysql database.However, if I try to look into the Subscriber table through Webmin, I do not see all the users but only a few. What should I do to reflect all the users through webmin as well?
Thank You, Padmaja ----- Original Message ----- From: users-request@lists.openser.org To: users@lists.openser.org Sent: Friday, November 02, 2007 3:27 PM Subject: Users Digest, Vol 30, Issue 2
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Today's Topics:
- Re: Set up two SIP to PSTN calls and then connect them (Andreas Granig)
- Traffic (Gerson A. Matiolli)
- Re: How to expose the expires value in REGISTER (Robert Dyck)
- Beep in audio stream. (Marc Dirix)
- Is it possible to insert avp to reply message? (Tung Tran)
- Re: Set up two SIP to PSTN calls and then connect them (Bogdan-Andrei Iancu)
- Re: Set up two SIP to PSTN calls and then connect them (CSB)
- Re: Is it possible to insert avp to reply message? (I?aki Baz Castillo)
- maddr in contact (Allan Chao ( ??? ))
- Re: Set up two SIP to PSTN calls and then connect them (I?aki Baz Castillo)
- Re: Traffic (Henning Westerholt)
Message: 1 Date: Thu, 01 Nov 2007 13:07:42 +0100 From: Andreas Granig agranig@sipwise.com Subject: Re: [OpenSER-Users] Set up two SIP to PSTN calls and then connect them To: CSB kjcsb@xnet.co.nz Cc: users@lists.openser.org Message-ID: 4729C18E.7040507@sipwise.com Content-Type: text/plain; charset=ISO-8859-1; format=flowed
It's already included, see http://svn.berlios.de/viewcvs/sems/trunk/apps/click2dial/
Andreas
CSB wrote:
Is there any update regarding the click2dial plugin that was planned to be introduced to the trunk?
Regards
Cameron
Message: 2 Date: Thu, 01 Nov 2007 10:35:49 -0200 From: "Gerson A. Matiolli" gerson@cambridgetelecom.com.br Subject: [OpenSER-Users] Traffic To: users@lists.openser.org Message-ID: 1193920549.5276.9.camel@jupiter2 Content-Type: text/plain
Hi, all
I am using Openser 1.2.2 - tls.
I have 400 registered users.
Everything works well as traffic is low.
If traffic is high, the calls are not completed (Busy tone)
Can anyone help me?
Message: 3 Date: Thu, 1 Nov 2007 09:12:20 -0700 From: Robert Dyck rob.dyck@telus.net Subject: Re: [OpenSER-Users] How to expose the expires value in REGISTER To: Christian Schlatter cs@unc.edu Cc: users@lists.openser.org Message-ID: 200711010912.20409.rob.dyck@telus.net Content-Type: text/plain; charset="iso-8859-1"
On Wednesday 31 October 2007, Christian Schlatter wrote:
Robert Dyck wrote:
I am wondering how to expose and test the value of the expires parameter in a REGISTER request.
I am experimenting with openser as the basis for a home phone network. I use multiple devices with the same user ID. They register locally ( with no reply ) and with an external service provider. The contacts are mangled to show the public address of openser. Multiple registrations result in a single AOR at the external registrar. Incoming calls from the outside are forked and ring the local phones. Local phones can also call each other without the hairpin problem associated with STUN enabled phones.
The problem is that a softphone will deregister when it is closed or its profile changes. This would deregister the AOR at the external registrar. The remaining phones could not receive calls from the outside until they refreshed their registrations.
I would like to prevent deregistration at the external registrar unless the phone that was deregistering was the only remaining one. The first step would be to identify REGISTER messages where the expires value is equal to zero.
Both 'Expires' header and 'expires' contact uri parameter have to be checked like e.g.
if ((is_present_hf("Expires") && $(hdr(Expires){s.int}) == 0) || ($(ct{param.value,expires}) == '0')) { # someone tries to unregister }
Have a look at http://www.openser.org/dokuwiki/doku.php/transformations:1.2.x if you're not familiar with the PV transformations introduced with 1.2.
I am indeed unfamiliar with PV transformations. I will have a look it. I was afraid I might have to do something ugly with regular expressions. I probably should not put off upgrading any longer.
Thanks, Rob
Message: 4 Date: Thu, 1 Nov 2007 21:35:06 +0100 From: Marc Dirix marc@electronics-design.nl Subject: [OpenSER-Users] Beep in audio stream. To: users@lists.openser.org Message-ID: 871BA93E-E51A-4297-906D-789BA5798461@electronics-design.nl Content-Type: text/plain; charset=US-ASCII; format=flowed; delsp=yes
Hi,
I'm currently setting op an openser server. My setup at the moment is as follows:
registrar (pstn) <=> yate (sip server) <=> openser <=> sip_phone.
As I make a call with the sip phone to a pstn line, the rtp stream is forwarded from the yate server to openser, which acts as en media proxy with rtpproxy. During the call I get very annoying beeps every 2 or 3 seconds.
The beeps sound a bit like cost-beeps or somethin.
When I connect the phone directly to the yate server however, which then starts acting as media-proxy, I do not get any beeps.
Furthermore, if I remove force_rtp_forward() from openser config, it stops being proxy for the stream, but still I get these annoying beeps. Excluding any problems with rtpproxy.
Clearly, the registrar sends these beeps, but he doesn't send them when I connect with yate. Am I missing something that can trigger this behaviour?
Thanks,
Marc Dirix
Message: 5 Date: Fri, 2 Nov 2007 09:41:33 +0700 From: Tung Tran tr.tung@gmail.com Subject: [OpenSER-Users] Is it possible to insert avp to reply message? To: users@lists.openser.org Message-ID: 200711294133.621270@VGN-TXN15P Content-Type: text/plain; charset="us-ascii"
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Message: 6 Date: Fri, 02 Nov 2007 05:30:45 +0200 From: Bogdan-Andrei Iancu bogdan@voice-system.ro Subject: Re: [OpenSER-Users] Set up two SIP to PSTN calls and then connect them To: Andreas Granig agranig@sipwise.com Cc: users@lists.openser.org Message-ID: 472A99E5.9000401@voice-system.ro Content-Type: text/plain; charset=ISO-8859-1; format=flowed
Hi,
Or you can use this script, with no external dependency:
http://openser.svn.sourceforge.net/viewvc/openser/branches/1.2/examples/web_...
regards, Bogdan
Andreas Granig wrote:
It's already included, see http://svn.berlios.de/viewcvs/sems/trunk/apps/click2dial/
Andreas
CSB wrote:
Is there any update regarding the click2dial plugin that was planned to be introduced to the trunk?
Regards
Cameron
Users mailing list Users@lists.openser.org http://lists.openser.org/cgi-bin/mailman/listinfo/users
Message: 7 Date: Fri, 2 Nov 2007 16:32:53 +1300 From: "CSB" kjcsb@xnet.co.nz Subject: Re: [OpenSER-Users] Set up two SIP to PSTN calls and then connect them To: "'Andreas Granig'" agranig@sipwise.com Cc: users@lists.openser.org Message-ID: 003401c81d01$0d9d8920$28d89b60$@co.nz Content-Type: text/plain; charset="us-ascii"
Thanks.
I currently use OpenSER and Asterisk and I can get the call set up using ctd.sh. The question I have relates to the accounting. Using the ctd.sh script is there a way to get the CDR records written from OpenSER? If I understand correctly, OpenSER drops out of the call signalling and will not receive any BYEs so accounting will be impossible; am I correct? Asterisk will record the calls but billing them appropriately using those records would be problematic (I think).
If using the SEMS option, am I correct in thinking that it would be possible to use the accounting records from OpenSER?
Regards
Cameron
-----Original Message----- From: Andreas Granig [mailto:agranig@sipwise.com] Sent: Friday, 2 November 2007 1:08 a.m. To: CSB Cc: users@lists.openser.org Subject: Re: Set up two SIP to PSTN calls and then connect them
It's already included, see http://svn.berlios.de/viewcvs/sems/trunk/apps/click2dial/
Andreas
CSB wrote:
Is there any update regarding the click2dial plugin that was planned to be introduced to the trunk?
Regards
Cameron
Message: 8 Date: Fri, 2 Nov 2007 09:39:31 +0100 From: I?aki Baz Castillo ibc@in.ilimit.es Subject: Re: [OpenSER-Users] Is it possible to insert avp to reply message? To: users@lists.openser.org Message-ID: 200711020939.31335.ibc@in.ilimit.es Content-Type: text/plain; charset="ISO-8859-1"
El Friday 02 November 2007 03:41:33 Tung Tran escribi?:
Hi all,
Please, when creating a **new** mail press "create new mail", but don't press "Reply" on any other mail of any other thread. If you do so your mail will appear contained in a wrong thread, broking it and make it very difficult to understand.
Thanks.
-- I?aki Baz Castillo ibc@in.ilimit.es
Message: 9 Date: Fri, 2 Nov 2007 17:15:45 +0800 From: Allan Chao ( ??? ) AllanChao@taiwanmobile.com Subject: [OpenSER-Users] maddr in contact To: users@lists.openser.org Message-ID: 970C4ACFBCFD2349B388E8907A2A7B5E80D2B4@TCCEXCH12.pcdc.com.tw Content-Type: text/plain; charset="big5"
Hi :
I have a flow UE -> gateway (use openser and runs proxy mode) ( ip : 192.168.1.2) -> SIP Proxy ( ip: 192.168.1.3),
if i have two user , UE1(192.168.1.5) and UE2(192.168.1.6) send REGISTER request to SIP proxy through gateway,
but our gateway add a maddr = "192.168.1.2" string in contact header,so the contact in REGISTER becomes <username@ host ; maddr="192.168.1.2">.
now , if UE1 send INVITE message to UE2, how does sip proxy to do if receive INVITE message? it will send invite message to UE2 through gateway ( maddr parameter) ? and does openser has support maddr in contact or not . thx.
allan
-
Padmaja