similar to: SNOM Hint for MeetMe

Dear all, I'm using Asterisk 1.4 as my SIP server of my LAN users. The SIP clients are using different operating systems such Debian, Gentoo and Windows XP so they use different SIP softphones like SJPhone, Twinkle and X-Lite. In order to let SIP clients to see the presence status to each other, do I have to establish any special setting in Asterisk 1.4 ??? Or the presence status (online,

Hi, I would like * to treat a SIP extension as a normal extension, when it comes to the busy functionality. In other words, if someone tries to call the SIP phone and there is already an ongoing conversation, the new caller should get a busy message/tone Is there any parameter that I can set? Is this something that should be configured at my softphone? Best, PHM

I've had my * setup installed with an X100P card for a couple of weeks and it's great fun! I'm even giving a demo to the local Linux group in a couple of days. But I have a snag. I have the X100P on a shared line, and configured to wait for 20 seconds before answering and doing the auto-attendant/voicemail dance. My problem is I can't find an application command to cancel the

I'm having trouble setting the outbound caller ID on calls I make from my PRI trunk group. The PRIs are served out of a 5ESS. Telco has set the PRIs up for user provided caller id information, so I believe I just don't have it set up right in my dialplan or something. I can't seem to find an example of setting the outbound caller ID specifically for a 5ESS. Does anyone have an

I'm running 1.2.9.1, and I can't get caller id dialplan matching to work. When calling from 9220370 to 1234, the following does not match. exten => 9220370/1234,1,NoOp(${CALLERIDNUM}) exten => 9220370/1234,2,Answer exten => 9220370/1234,3,Playback(tt-weasels) However, when calling from 9220370 to 1234, this DOES match. exten => 1234,1,NoOp(${CALLERIDNUM}) exten =>

I am struggling to find out how to code hint in AEL2. I did hint(Custom:light1) and it keeps complaining about the : (colon). It works fine for SIP device like hint(SIP/439). Anyone who has tried it before?

PH> I am setting up a snom 360, and the lights come on OK when the mapped PH> user makes an outgoing call, but when the user takes an incoming call PH> the light does not come on. PH> I do not want to install the bristuff patch if possible. PH> (although I can see that with the devstate command I can make the lights PH> do whatever I want) First, ensure that the 360 has

Sorry for posting this again, but it seems to have become attached to another thread. Guess I replied to another message instead of starting a new one... Hi, I'm trying to setup a call forwarding rule so that when an extention doesn't answer the call is forwarded to my mobile. I'm using voiptalk.org for incoming and outgoing calls and SIP phones for extentions (so all IP based -

Has anyone seen this before, and know what is happening? USER at HOST:~/asterisk/agx-ast-addons# ./build.sh -- Configuring done -- Generating done -- Build files have been written to: /root/asterisk/agx-ast-addons [ 11%] Building C object CMakeFiles/app_devstate.dir/app_devstate.o Linking C shared module dist/app_devstate.so [ 11%] Built target app_devstate [ 22%] Building C object

Hi I am trying to get a SNOM 360 to monitor other extensions i.e. when someone makes a call to/from another extension, one of the LED's on the SNOM 360 will change state. I am using 1.0.9/bristuff-8l. I have 2 extensions - 2001 is a SNOM 190, 2002 is a 360 - both are running the latest firmware. (3.60i for 190, 4.0 for 360). I have read all the relevant articles on the wiki on

Ok, I'm tearing my hair out trying to get Asterisk moh streaming to work. After several hours jerking around with icecast and muse, I tried to point my asterisk system directly at two streams I know work. This is what extensions.conf has: [default] mode=quietmp3 directory=/var/lib/asterisk/mohmp3 [stream2] mode=custom directory=/var/lib/asterisk/mohmp3-empty

Apologies if this is a resend, but I've sent this 12 hours ago and still can't see it on the list. Hi, I've just started to setup my phones with Buddy watch. Basically, it all works fine when using the simple example on the wiki: exten => 123,hint, SIP/some_sip_reg exten => 123,1,SIP/some_sip_reg BUT, what I need to do is dynamically decide where the hint checks for buddy

I've got hints setup for my MeetMe conferences like so: exten => _60X,hint,MeetMe:${EXTEN} and they show up in "core show hints" like so 600 at dialtone : MeetMe:600 State:Unavailable Watchers 1 _60X at dialtone : MeetMe:${EXTEN} State:Unavailable Watchers 0 I'm wondering why they're Unavailable instead of

Hi, With 0.0.6pre3: # ./build.sh CMake Warning (dev) in CMakeLists.txt: No cmake_minimum_required command is present. A line of code such as cmake_minimum_required(VERSION 2.6) should be added at the top of the file. The version specified may be lower if you wish to support older CMake versions for this project. For more information run "cmake --help-policy CMP0000".

Hi all, I've programmed on a SNOM function key the destination "**77" which in my extension plan reprograms/toggles an asterisk DB variable which I use in the extension plan for some call routing decisions. I would like the SNOM extension light to permanently reflect the current toggle status of my application logic/asterisk DB variable. Is something like that possible to do in

Hello Happy New Year to all!! I've just completed porting from Asterisk 1.2 to 1.4. I did this by doing a clean install on a new box, and moving over configuration and scripts where needed. All went surprisingly well! Anyway, one lingering issue is that the function key "lamps" on our Snom phones have all stopped working! We're using a mix of Snom 290/320/360 phones and

Hello All! Just a notice that our PHP/Smarty-based GPL version of PhoneCALL version 2.6.1 has been released, and is the current stable release. http://www.vecsector.com/phonecall We're always looking for feedback/testers to help us enhance it and make it even easier for everyone to use. The current version is designed around the advanced Asterisk user, and we are working on a more

Hello list members, after programing of dialplan I have some messy Custom:hints which I can see in 'devstate list'. I didn't find any possibility how to remove this hints from Asterisk and I want remove them.? Can you help me with that, please? I tried search about that something in documentation or on Google, but I didn't find anything.? asterisk*CLI> devstate list ?

Hello all! It's come time where I need to add another programmer to our team. You should have at least 3 years of "work" experience with PHP/MySQL. Please send me your resume and a few code samples if you can. If you can only work part-time or full-time, please include that in your response. Along with your salary requirements. You'll be working with PhoneCALL, so be sure to

Does anyone have an example of hoe to use these two commands? I have read he documentation, and I am still unclear on where this command goes, as part of extensions.conf or where? If someone could post a working example it would be most helpful. Regards to all, Joe