similar to: Remote-Party-ID implementation on trunk

Hi, i have the same problem. Trying to use the dialplan function CONNECTEDLINE() this way Set(CONNECTEDLINE(name)=${SIPPEER(${EXTEN},callerid_name)}) Set(CONNECTEDLINE(num)=${EXTEN}) ends with [Jul 1 16:30:50] ERROR[3954]: pbx.c:2902 ast_func_write: Function CONNECTEDLINE not registered Same happens trying function CALLEDID. I am using Asterisk 1.6.1.20. What do i

Hello, I am using a Cisco 2,811 gateway to connect Asterisk over PRI to our Nortel TX-1 PBX. Up to now I had only the calling party names passed both ways. After upgrading the Cisco to the 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

Hello, 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 way...)? BTW, one note to the above patch: To make it work the device should have the parameter sendrpid set to true.

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:

>>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 way...)? Asnwering myself: I am using realtime extensions, so I've added call to

We've been using Asterisk 16 for a while now, and tried turning on send_rpid = yes in my pjsip config for end points.  This solves a problem we're having where attended transfers aren't updating the CallerID when the transfer is complete (it would show the callerID of the party attempting the transfer, and never update after the transfer happened). The side effect of this change

Hello, I am running a small installation of asterisk and looking for future expansion of it to handle thousands of users. From what I read I see that usually large installation place OpenSER (or similar solution) in front of Asterisk in order to provide high call rate because "OpenSER does only signalling while Asterisk does all". My question is: If Asterisk also does only signalling

> 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 feature. I guess I need some >> conference server; any experience

Hello, I see that most people are using the extensions.conf 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:

Hello, I would like to implement a few decision making process inside the dialplan using information stored in MySQL (like LCR, etc.). 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?

Hello, Our university has to upgrade soon its old Nortel PBX's which holds around 10,000 extensions tied to 5 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?

I don't expect my SIP provider to provide useful "Remote-Party-ID" information. Therefore, I am using "CONNECTEDLINE(num)=xxx" AND "CONNECTEDLINE(name)=yyy" to populate remote party information from a local database. I am also using the "I" (upper case "i") option for Dial. Generally I like to see to see the remote party name appear on the

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

Hello, I tried a few months ago to use IMAP with Asterisk; I used either 1.4 or the latest SVN at that time (sorry, don't remember). After a day I had to remove it since Asterisk crashed, mostly in the IMAP client code (the code of UW IMAP). My users wants IMAP back (they loved it) but not in the price of crash... I could not reproduce the crashes at the lab. They only occour on the

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:

Hello, I am having an issue here that after an attended call transfer there is no audio on one way; the problem is caused by Asterisk sending two INVITE messages without waiting for an ack for the first one. The issue has been reported on bug 9305, has been fixed and the fix is now included inside the main stream (version 1.4.21). However, I still get this behaviour, so I opened a new bug

Hello, When dialling a SIP 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:

When I set CONNECTEDLINE() info for an incoming ISDN call, the calling party sees only CONNECTEDLINE(num) and the name does not get displayed. Some time ago I called a number, where I did get back a name and a number and everything was displayed correctly. So I think the calling site should basically be able to handle all connected line info. Looking at a pcap trace of the D-channel data, I

We have a scenario where we wish to present a toll-free caller id, yet have our calls rated based on our billing-telephone-number. Is it possible to present a number in the sip header for billing and another number in the header for jurisdicional call rating? Whereas today, all of our calls are billed at the highest rate (intra-state) because we're presenting a number that isn't in the

Hello ppl, Sorry to re-post it, but kinda these issues are getting on my nerves. I tried Set(CALLERID(num)=7329) on 1.2.12, which works fine, but not on 1.4.4. The problem : 1. I receive call from caller 'AAA' on my number, 'BBB' which is on my Asterisk box. 2. I have to redirect the call to some other number, say, my cell num - 'CCC'. 3. My PSTN provider wants the