Robert Rozman
2005-Oct-23 01:30 UTC
[Asterisk-Users] Anyone using Java SIP communicator with Asterisk ?
Hi, this java video softphone claims it can operate with Windows messenger. It's also mentioned on this web page http://www.voip-info.org/wiki/view/SIP+COMMUNICATOR But I couldn't find any more info on how to set it up with Asterisk and how compatible is with other video softphones... Anyone with such experience or working installation ? Thanks in advance, regards, Rob.
Mohamed A. Gombolaty
2005-Oct-23 02:42 UTC
[Asterisk-Users] Call Admission Control in Asterisk
Dear All, I was trying to limit the number of calls between different located sites in order to avoid congestion of the bandwidth, but as I found from the mails and testing that it is easy to do it for the incoming calls by the setgroup() and group_count while it is the outgoing is hard to track or limit, So I was wondering if we will see a Call Admission Control soon in Asterisk that can do this job or not? Thx MAG
trixter aka Bret McDanel
2005-Oct-23 02:54 UTC
[Asterisk-Users] Call Admission Control in Asterisk
On Sun, 2005-10-23 at 11:42 +0200, Mohamed A. Gombolaty wrote:> Dear All, > > I was trying to limit the number of calls between different located sites in > order to avoid congestion of the bandwidth, but as I found from the mails and > testing that it is easy to do it for the incoming calls by the setgroup() and > group_count while it is the outgoing is hard to track or limit, So I was > wondering if we will see a Call Admission Control soon in Asterisk that can do > this job or not?setgroup() should work for outbound. Did you try it and have problems? In asterisk 1.0.x there is a bug about transfered calls, is that where you were running into problems? I find this unlikely since you referenced group_count, which is a 1.2 function (replacing the deprecated checkgroup()). http://www.voip-info.org/wiki-Asterisk+cmd+SetGroup -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 189 bytes Desc: This is a digitally signed message part Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20051023/2f6d8afc/attachment.pgp
gwynpen
2005-Oct-25 01:03 UTC
[Asterisk-Users] Anyone using Java SIP communicator with Asterisk ?
Hi, I gave it a quick try (audio only): - set Public SIPaddress, SIP registrar, SIP-proxy etc. to the IP of the asterisk - set DEFAULT_AUTHENTIC... to 'asterisk' - removed all STUN entries etc. - provided user name and pwd according to configured SIP friend -> SIP communicator registers with asterisk outgoing/incoming call -> signalling seems to work, but no audio due to difficulties to find the appropriate codec. I'll give it another try later on... Cheers J?rg> -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of > Robert Rozman > Sent: Sunday, October 23, 2005 10:30 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [Asterisk-Users] Anyone using Java SIP communicator > with Asterisk ? > > Hi, > > this java video softphone claims it can operate with Windows > messenger. It's also mentioned on this web page > > > http://www.voip-info.org/wiki/view/SIP+COMMUNICATOR > > But I couldn't find any more info on how to set it up with > Asterisk and how > compatible is with other video softphones... > > Anyone with such experience or working installation ? > > > Thanks in advance, > > regards, > > Rob. > > _______________________________________________ > --Bandwidth and Colocation sponsored by Easynews.com -- > > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >