2 Nov
2007
2 Nov
'07
1:34 p.m.
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(a)lists.openser.org>
To: <users(a)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:
>
> 1. Re: Set up two SIP to PSTN calls and then connect them
> (Andreas Granig)
> 2. Traffic (Gerson A. Matiolli)
> 3. Re: How to expose the expires value in REGISTER (Robert Dyck)
> 4. Beep in audio stream. (Marc Dirix)
> 5. Is it possible to insert avp to reply message? (Tung Tran)
> 6. Re: Set up two SIP to PSTN calls and then connect them
> (Bogdan-Andrei Iancu)
> 7. Re: Set up two SIP to PSTN calls and then connect them (CSB)
> 8. Re: Is it possible to insert avp to reply message?
> (I?aki Baz Castillo)
> 9. maddr in contact (Allan Chao ( ??? ))
> 10. Re: Set up two SIP to PSTN calls and then connect them
> (I?aki Baz Castillo)
> 11. Re: Traffic (Henning Westerholt)
>
>
> ----------------------------------------------------------------------
>
> Message: 1
> Date: Thu, 01 Nov 2007 13:07:42 +0100
> From: Andreas Granig <agranig(a)sipwise.com>
> Subject: Re: [OpenSER-Users] Set up two SIP to PSTN calls and then
> connect them
> To: CSB <kjcsb(a)xnet.co.nz>
> Cc: users(a)lists.openser.org
> Message-ID: <4729C18E.7040507(a)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(a)cambridgetelecom.com.br>
> Subject: [OpenSER-Users] Traffic
> To: users(a)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(a)telus.net>
> Subject: Re: [OpenSER-Users] How to expose the expires value in
> REGISTER
> To: Christian Schlatter <cs(a)unc.edu>
> Cc: users(a)lists.openser.org
> Message-ID: <200711010912.20409.rob.dyck(a)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(a)electronics-design.nl>
> Subject: [OpenSER-Users] Beep in audio stream.
> To: users(a)lists.openser.org
> Message-ID:
> <871BA93E-E51A-4297-906D-789BA5798461(a)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(a)gmail.com>
> Subject: [OpenSER-Users] Is it possible to insert avp to reply
> message?
> To: <users(a)lists.openser.org>
> Message-ID: <200711294133.621270@VGN-TXN15P>
> Content-Type: text/plain; charset="us-ascii"
>
> An HTML attachment was scrubbed...
> URL:
>
http://lists.openser.org/pipermail/users/attachments/20071102/26276838/atta…
>
> ------------------------------
>
> Message: 6
> Date: Fri, 02 Nov 2007 05:30:45 +0200
> From: Bogdan-Andrei Iancu <bogdan(a)voice-system.ro>
> Subject: Re: [OpenSER-Users] Set up two SIP to PSTN calls and then
> connect them
> To: Andreas Granig <agranig(a)sipwise.com>
> Cc: users(a)lists.openser.org
> Message-ID: <472A99E5.9000401(a)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(a)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(a)xnet.co.nz>
> Subject: Re: [OpenSER-Users] Set up two SIP to PSTN calls and then
> connect them
> To: "'Andreas Granig'" <agranig(a)sipwise.com>
> Cc: users(a)lists.openser.org
> Message-ID: <003401c81d01$0d9d8920$28d89b60$(a)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(a)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(a)in.ilimit.es>
> Subject: Re: [OpenSER-Users] Is it possible to insert avp to reply
> message?
> To: users(a)lists.openser.org
> Message-ID: <200711020939.31335.ibc(a)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(a)in.ilimit.es
>
>
>
> ------------------------------
>
> Message: 9
> Date: Fri, 2 Nov 2007 17:15:45 +0800
> From: Allan Chao ( ??? ) <AllanChao(a)taiwanmobile.com>
> Subject: [OpenSER-Users] maddr in contact
> To: <users(a)lists.openser.org>
> Message-ID:
> <970C4ACFBCFD2349B388E8907A2A7B5E80D2B4(a)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
>