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?
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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"
An HTML attachment was scrubbed...
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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
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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