search for: yehavi

Hello, One of our faculties have Microsoft's LCS and would like to connect it to our Asterisk system. the problem is that Asterisk talks SIP over UDP while LCS talks SIP over TCP with TLS. Anyone can recommend a gateway between these two protocols? Thanks! __Yehavi:

> On Sun, 18 Mar 2007, Yehavi Bourvine +972-8-9489444 wrote: > >> Hello, >> >> >> On most SIP phones a conference call is done on the phone and is limited to 3 >> participants. Polycom phones has a configuration option to use a conference >> server instead of the internal conferencing fea...

...BXes. Up to now we thought about commercial solutions but now there is a window openning for open source solution. However, I need examples to convince that this solution is feasible, and preferably at other universities. Are there any pointers for such installations? Thanks! __Yehavi: -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081121/c4ddec4a/attachment.htm

Hello, I am using the SetCIDNum dialplan application on 1.2 and 1.4.0; I've tried it on 1.4svn 56126 and it does not recognise this application. Any idea?... Thanks! __Yehavi:

...f syntax (most of the examples and questions here use that syntax). recently I've translated all my dial plan to AEL syntax and I find it much easier, especially when you need IFs. Why most people don't use it? Am I missing something? Thanks! __Yehavi:

...ng while Asterisk does all". My question is: If Asterisk also does only signalling (i.e. the voice traffic goes directly between the phones and not via asterisk) is it still that slow? I preffer to have one software package rather than dealing with two. Thanks! __Yehavi: -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081118/6a89d006/attachment.htm

Use Snom phones. We have had around 6 participants, without problems. In theory you should be able to have around 12 people on a conference on a snom phone. Jon -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Yehavi Bourvine +972-8-9489444 Sent: 19. marts 2007 09:14 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Conference server (or how to make a call withmore than 3 u > On Sun, 18 Mar 2007, Yehavi Bourvine +972-8-9489444 wrote: > >> Hello, >> >> >> On most S...

...I see the MYSQL() application, but as far as I understand I have to connect to the database each time I want to query it; this seems a CPU eater to me. Is this indeed the case, or can I open it once Asterisk starts and leave it open? Thanks, __Yehavi:

...ork with a static data. Where do I take the name of the called person (the "equivalent" of CALLERID, but the other way...)? BTW, one note to the above patch: To make it work the device should have the parameter sendrpid set to true. Thanks, __Yehavi:

...e system with users. I suggest to write the IMAP client code by the Asterisk developers and not depend on external code. In any case, I'll try this week to upgrade to 1.4.6 version and then add IMAP support and inform what happens. Thanks! __Yehavi:

...d inside the main stream (version 1.4.21). However, I still get this behaviour, so I opened a new bug (13120). This bug sits there for over a week with no reponse... Has anyone else noticed this behaviour? Any idea what I can do? My users are angry on me... Thanks! __Yehavi:

...latest release (12.4.24T) it began honoring the "remote-part-ID" field sent by Asterisk and sends the *called*name to the Nortel. However, I still do not get the called name from the Nortel to Asterisk. Has anyone managed to make this working? Thanks! __Yehavi: -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090607/0b370c9a/attachment.htm

Hello. I'm trying to use gmail's imap feature w/ asterisk imap voicemail. I compiled c-client with the following settings: make lr5 IP6=4 and asterisk with: ./configure --with-imap=/usr/src/imap-2007a/ However if i enable any if the imap settings in voicemail.conf, asterisk starts acting funny and dosent allow any calls imapserver=imap.gmail.com imapport=993 mapfolder=Voicemail Where

Hi. Has anyone had any success in implementing a callback or ringback function in Asterisk? I've had a look at the callback-voicemail example on voip-info.org http://www.voip-info.org/wiki/view/Asterisk+tips+callback However it won't quite work for me. I need it for local SIP users which most of them don't have voicemail. If one SIP user calls another SIP user and the second user is

What I would like to do is have two identical * servers which accept registrations of sip extensions 4000-4999. If I define a rrDNS or LinuxHA then I should have load-balanced registrations. However, say ext. 4001 is registered on *1 and 4002 is registered on *2, if 4001 tries to call 4002 then I would like to do something like: - lookup 4002 on *1, try to establish a call if it's

...left "in use" and could not receive new calls. - After seeing reference to similar problem on this list I;ve downloaded today the latest SVN source code and installed it. The problem is that it shows the call limit as 0 and not as 1. Any idea? Thanks, __yehavi:

>>Yehavi wrote: >> > I am trying to apply the "called party identification" >> > patch (patch 8824) and managed to make it work with a >> > static data. Where do I take the name of the called person >> > (the "equivalent" of CALLERID, but the other wa...

...phone which is already in a call the caller hears a "regular" ringing tone and does not know that the called party is engaged in another call. Is there a supported way inside SIP to tell the calling party to play a stuttered ringing tone? Thanks! __Yehavi:

Hello, I've found that my Polycom-501 accepts INVITES from any server in the world... I would like to restrict it to accept calls only from the servers listed in its config file, but I cannot find anything in the documentation. Any idea? Thanks, __Yehavi:

...phone? (define extension 3 on both doesn't work as only one can register with it). What should sla.conf file have? Do I have to change extensions.conf? (To make it simple let's assume that it contains only Dial(SIP/${EXTEN}) as the dialplan). Thanks! __Yehavi: