similar to: SNOM phones stay "in use" after transfer

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 having some troubles with Snom phones and maybe someone can help me. Let me say this: BLF and pickup works great with Polycomes and Grandstream etc... So I think my problem might not be Asterisk related but I am not 100% sure. The snom phones subscribe to my extensions (hint priority) as expected. The light blinks (ringing) or is turned on (in the call) as expected. My problem is to

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 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

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 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 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 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 am running Asterisk 1.4 with the RPID patch (8824) in order to display the name of the *called* person. It works on Cisco and Polycom phones, but not on SNOM. The local SNOM dealer claims that he saw it working, but he cannot give me more details. Has anyone managed to make it working? Thanks, __Yehavi: -------------- next part -------------- An

> 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 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

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, I found some issues with Snom 320 message retrieve key. This button works only when the MWI does not blink! If MWI blinks and I do press retrieve button I get "Unknown" on display and busy tone. From the sip debug it looks like that Snom does not send credentials to Asterisk which responds with 407 Proxy Auth required. I have loaded Snom with latest 5 firmware. No change. I'm

Hi! I have the following questions/problems with * 1.4. We have several Snom phones (320 and 360). Hints are configured in extensions.conf (core show hints shows the correct values). My Snom phone is registered to some numbers (validated by using sip show subscriptions). I see the lights blinking if someone calls the subscribed number and steady lights if the call is established. So far, so

I have setup an asterisk based phone system using snom-320 (SIP based) phones. I would like to change what seems to be the default procedure for an attended call transfer. Right now, the phone user places the call on hold, calls the extension using a "extension" button on the phone, speaks with the call recipient , and presses transfer to transfer the held call. The users would like

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

It's very possible that the Polycom IP600 will work with this. As it is just an implementation of a SIP standard for subscribing to the state of other extensions. As for the feature improvements you might see them from me, but not very likely. It is easier for me to train my customers to hit *8 (I will probably just program a pickup button for them) than it is for me to figure out what I

I concur with Conrad. Cisco phones were retrofitted for SIP, whereas Snom phone are built around, and expressly for, the SIP standard. To be in compliance with Cisco regs, you are also supposed to have a SIP User license and a Smartnet contract for each phone if you abide by their program. I realize it is not difficult to obtain the SIP firmware, but if you were dealing with a Cisco authorized

Hi all, One of my users reports frequently hearing echo on her Snom 360 phone, even while talking to other Snom phones (via Asterisk) on the same LAN (i.e., all-digital low-latency connection). I can never reproduce it though, and swapping the phone didn't help. Has anyone else seen "mystery echo" on Snom phones? Any suggestions for debugging? On my own Snom 360, I sometimes