similar to: PRI/Q.sig between Cisco & Nortel

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, 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 am using a Cisco-2,811 router with PRI as a gateway between Asterisk and Nortel TX-1. I had problems with name transfer and with the help of Cisco support I've fixed it. Enclosed here are the definitions needed for it. BTW, Cisco's CCM is using MGCP thus the Q.sig is handled by CCM. Here I am using SIP so the router must decode/encode the Q.sig. The Nortel should be defined

> 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?

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, 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?

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 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, 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:

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

Hello, I am looking for a "real life" example of using SLA lines under Asterisk. I'll describe my environment and would like to know how I define it in Asterisk (version 1.4 final). Suppose I have two multi lines phones. The first phone has extension 1 assigned to it, and the second phone has extension 2 assigned to it. Now, I want extension 3 to be available on both phones as

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

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

I've been trying to look up information on upgrading firmware on a nortel i2004 ip phone. I have this phone leftover from a trial, and it's supposed to be upgradable to current firmwares. Since I also run a DMS I was able to login to nortel's site and get all the firmware files, but All the NTP's regarding firmware upgrading these are how to tell you BCM to send the file to it.