search for: __yehavi

...4 isdn incoming-voice voice isdn supp-service name calling (This receives names via Q.sig) isdn negotiate-bchan isdn outgoing ie facility isdn outgoing ie caller-number isdn outgoing ie called-number no cdp enable Anc the rest is quite standard. Regards, __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:

...). 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:

...IP 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:

...ling 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

...he conference call to call the others and join them by herself. If I understand correctly, with the MeetMe you have to initialize the conference and then people dial by themselves into it. This won't be acceptable by the secretaries here... Thanks, __Yehavi:

...he conference call to call the others and join them by herself. If I understand correctly, with the MeetMe you have to initialize the conference and then people dial by themselves into it. This won't be acceptable by the secretaries here... Thanks, __Yehavi: _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

...onf 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:

...ive 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:

...ded 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:

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

I setup a number of remote phones on public IPs using the web interface. Now my question is how do I change the default Polycom:456 password via the web interface. Is there a hidden way or does it have to be done via FTP TFTP? Thanks, Steve Totaro

...e 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:

...PBXes. 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

Hi, What is the best way to implement Automatic Redial on No Answer ? Looking at http://www.ietf.org/internet-drafts/draft-ietf-sipping-service-examples-12.txtI can see how Automatic Redial on Busy could (should) be done. How would you do it on No Answer ? Is there any event you should SUBSCRIBE to so that you're notified that you're callee is available ? What if you ask to be notified

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

...protocol forces the Cisco to use ESGF signalling). We could not use H.323 as it forces the Cisco to use ISGF. I suspect that SIP is the same, but setting ISGF signalling on Nortel doesn't help. Anyone had some luck with this configuration? Thanks, __Yehavi:

Can anyone point me to info on how to change the functionality of the SIP (7.4) 7960's. We previously had an SCCP firmware on the phone and the users want the phone to work like it used too. Here are some examples: The users do not want to push the new call softkey or the speaker button in order to dial a call. They want to be able to just begin dialing the number. The users do not want

...hold" state forever, and Asterisk won't accept new calls to it since it has max calls of 1... Similar thing happens with attended transfer (putting the call on hold, calling the other party and either disconnecting or pressing Transfer). Any idea? Thanks! __Yehavi: BTW, any idea when 1.4.1 is scheduled to be released? I need the SLA feature desperately...

I have a handful of 7912's connected to my asterisk 1.2.14 server. (6 to be exact). I get the X on the display sometimes for loosing registration. I have the config file for the 7912's SipRegInterval: 60 and asterisk is the default. ; maxexpirey=3600 ;defaultexpirey=120 I've not changed them. How can I keep these phones online and stop loosing registration? Thanks, Jerry