similar to: Configs for IAX <> IAX trunk

Lets say I have an * at my business, with 7960 SIP phones. All the sip phones are registered using their extension number (like 305), but I would also like to put my SIP URI on my business card and in a name format, not an extension number (like lee.goodman), so that the SIP URI would read lee.goodman@asterisk.company.com. How would I set this up in extensions.conf? I got

Has anyone used Gastman/Astman successfully? I have it up and running (Gastman win32), but have a problem with the creation of end stations on the map. I'm not sure of the format of the extension to use when creating a end station icon. Services like Conference bridge and Musichonhold seem to work ok (I use 555@mainmenu and 666@mainmenu) for the Icon extensions. IAX softphone seems to work

So I setup the MeetMe application in Asterisk Assigned an extension to it. When one of my SIP phone dials the conference extension, they get a message "you are the first one in the conference", so far so good. When the 2nd SIP phone dials the conference extension, they get a busy signal Now I know that you have to have a Zapta device to enable conference application. I have an X100P (1

Hello, Hope everyone is enjoying their holiday! We setup two asterisk servers (From CVS on Wednesday) and set up IAX between the two. Right now they both reside on a switch with a static 192.168.0.x IP address. The first Server is .5 and the second is .30. Our dialplan seems to be working, however on the console we get a flurry of NOTICE and WARNING messages. NOTICE[1116941120]: File

I have an inbound and outbound account with Voicepulse (I am very happy with the service, btw). But I have a weird IAX2 problem. When I get a inbound call on my Voicepulse DID, the call hits my asterisk server correctly with the correct callerid (the DID phone number 617902xxxx). when the call gets passed on to a softphone (X-lite), the caller id that shows up on the X-lite softphone as Lee ,

I'm having a problem with IAX running through an intermediate asterisk box. Perhaps a small diagram will explain the situation better: *A ------- [cloud (public internet)] ------- *B --------[cloud (private network)]----------- *C Asterisk server's A, B, and C, are all connected together with IAX All asterisk servers are 1.6.0.6 Server A and B are geographically close, but connected over

as i said, right now i'm just getting my feet wet. but, i will be needing to build dialplans on the fly. 'add extension' seems like the right call to make. .t > What is the goal of this? It doesn't make much sense to me. Care to > share some insite into what your goal is? > > bkw > > On Sun, 23 Nov 2003, tad wrote: > > > actually, i do have a

Is there any documentation on configuring IAX between * machines? I've noticed references to many topics in the config files, including: - dialplans - trunking - authentication - transfers But before I go and try to grok 8000 lines of source (in one file, no less) I was hoping that somewhere there exists even something like a man page that describes the configuration options.

I've built the latest CVS of asterisk -- not the zaptel or libpri directories, just the asterisk directory. asterisk installs successfully, but there are severe problems. I built this system in the past and ran it, but now building it again fails. This is the CVS as of this morning, 2003-06-13, but I had problems on 06-11/12 as well. After make; make install; make samples; make config, I

Can anyone explain why this is happening? I have a server attached to a phone line that will play a .wav file, then play all the dtmf digits (after it answers the call). If I place a call from a SIP device (like a Cisco 7960 phone) through Asterisk and on to the test server, via PSTN, the .wav file sounds fine, but the DTMF digits are distorted ----->------------->--------------------audio

Does the Asterisk server support VAD (aka Silence Suppression)? I want my SIP phones (7960's) to use VAD when dialing out the x100P interface. I know the phone can do VAD , can the Asterisk server be setup to do it? and if so, where do I set the configuration? Thanks Lee Goodman -------------- next part -------------- An HTML attachment was scrubbed... URL:

I've seen this happen a few times and I think it's when the system that Asterisk is running on crashes due to a power failure (or for some other reason that causes a non-planned shutdown). While Linux comes up fine, Asterisk won't start because the drivers are loading in the wrong order. fixed by: 1) sh /usr/src/fix-asterisk-modules.sh 2) sh /etc/init.d/asterisk start Is this a

Hi We are trying to get better sound quality out of the prompts on our Asterisk system. We had some new ones made by thevoice.digium.com and they are in WAV format instead of the default GSM format on the Asterisk server. The problem is, when you dial in to the server using the FXO card (X100P) you really can't tell the difference between the WAV prompt and the GSM prompt, however, if you

This article below came up on the newwire. The RTCP XR RFC was published. Will Asterisk be supporting this function in a future release? Does anyone know if any phone vendors are going to be supporting it? Thanks Lee Goodman Our Technology Update this week is about one of those mechanisms. Known as RTP Control Protocol Reporting Extensions (RTCP XR), the technology defines a standard way to

On Aug 7, 2007 'Mojo' wrote: Nicholas Blasgen wrote: > I've got 4 SIP phone lines with a call-limit of 2 for each. I've > written a handy macro to allow my users to dial a phone number and the > macro will figure out the next available line to use by first checking > if the GROUP() is over 2 and then checking to see if ChanIsAvail() as a > backup, and if it

I have tried for over a month off and on to get iaxtel for inbound to work... and tonight after alot of troubleshooting we noticed this: iaxtel inbound will use the last entry in your iax.conf to auth against. So if [iaxtel] is at the top and say [voicepulse] at the bottom. An inbound call will try to auth against that [voicepulse] entry even with the [iaxtel] entry at the top of the file. Has

Hi Lee, You need to specify the VM context that you are using.. so using your examples.. extensions.conf entry.. exten => 1000,1,Dial(SIP/1000,20) exten => 1000,2,Voicemail2(u1000) exten => 1000,102,Voicemail2(b1000) exten => 1000,103,Hangup should be.. exten => 1000,1,Dial(SIP/1000,20) exten => 1000,2,Voicemail2(u1000@sip) exten => 1000,102,Voicemail2(b1000@sip) exten

Hello, One quick question. Does anyone has experience implementing unified messaging (UM) using Asterisk. Does Asterisk has support for UM ? Thanks, Tarun ___________________________________________________ Medicine meets Marketing; Dr. Swati Weds Jayaram. Rediff Matchmaker strikes another interesting match !! Visit http://matchmaker.rediff.com?2

I have all my SIP users in a realtime database. I would like to use MySQL command to query the database and use the results from the query to page all the phones found in the query. The results from the MySQL query will be multiple rows of extension: Something like: mysql> Select extension from sip where extension like '6%' 6001 6002 6003 ex.... I need to put all the results into a

Hi all, I have a question regarding using T38 for fax sending and here is my scenario: fax -> SIP ATA (T38 enabled) -> Asterisk #1 -> IAX TRUNK -> Asterisk #2 -> SIP ATA (T38 enabled) -> fax My question is, how can I know if I'm really using T38? is T38 information coming to the other side (because of SIP to IAX conversion) or just plain g711a data? I'm using Linksys