similar to: SetCIDNum is not available on 1.4svn

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:

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, 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

> 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

I have achieved much of what I wanted to achieve using Wine, with one exception. I was hoping to be able to use Shockwave content, but despite installing Firefox and the Shockwave plugin, it does not work - it seems to stop after the promotional film before the actual requested content. I know there are "pay" solutions to this problem, but was wondering if anyone had found a free

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. 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

Dear all, I tried to make a call with extensions.conf. exten=> _00[1-9].,1,Dial(zap/g1/${EXTEN}) exten=> _00[1-9].,2,NoOP(ANSWEREDTIME=${ANSWEREDTIME}) exten=> _00[1-9].,102,Hangup But the 2 and 102 will not be executed. So I can get the correct answered time via 2. Is any idea about it? Is it the problem of my ZAP channel's configuration? My zapata.conf is as below:

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

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?

Hi, I am using asterisk 1.4.18. I am using it for inbound only call center. The SIP phones are X-Lite. Right now when a call is proxied by Asterisk to X-Lite the agent only sees asterisk written on its CLI screen. I want the agents to be able to view the callees number. Is there any way to make this happen. Regards Syed Nasruddin -------------- next part -------------- An HTML

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 running version 1.4 with realtime support. I've set (for Snom phones 300/320/360) a call limit of 1 (incominglimit and outgoinglimit fields in the database). - When I used 1.4 SIP SHOW PEER show that it has a call limit of 1. The problem was that when such a phone received a call and did attended transfer it was left "in use" and could not receive new calls. -

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:

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:

I am trying to install RepliWeb R-1 Console, but it is failing. I have a self-extracting installer that I used on my Windows-based laptop but when I try to run that with Wine, this happens: I do: wine R1_win.exe Some extraction happens, then: File not found C:\windows\temp\rw_products>.\check_before_install.exe err:ole:CoGetClassObject class {88d969c0-f192-11d4-a65f-0040963251e5} not

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

A few months ago, I tried to get Datakam Player (Registrator Viewer) to work on my Linux system using wine, without success. I have since installed a newer wine, and it now starts and can open the files created by my dashcam, but it does not play them - the video area remains black, there is no sound and the speed and location displays do not change. Short of trying an even newer wine, which

I am now at the point where I have one more thing to install before I can scrap my Windows box - and it's Leisure Suit Larry's Greatest Hits and Misses. I have installed the DOS-based parts of it in dosbox, and also used sarien and freesci to handle some of it, but there is a significant part of this collection that needs Windows. Trying an installation with the default Windows version