similar to: Dynamic feature support recently added to CVS HEAD

Anybody ever put a Sangoma and a Digium card in the same server? Specifically a four port card from each company? -bill wlloyd@slap.net

Scenario is as follows. Caller comes in over ZAP channel connects to handset on another ZAP channel. Call is bridged. I'd like the callee to be able to hangup on the caller and then be presented with a agi application. Basically the agent that answered the call has to enter a few responses to questions asterisk asks. On some ACD phone systems this is called a "wrap code".

I too have been having SIP problems the last couple of days, I get the same message as Louis-David but in this setup only: PSTN >-ISDN-> AS5300 >-SIP-> * I can make outbound calls no problem but inbound calls seem to stall, according to 'sip debug' it just says 'Ignoring this request' but I cant establish why .... > -----Original Message----- > From: William

I'm in search of useful ACD type statistics from the queues. Ie talk time, ratio's, dropped calls etc. The flat file queue_log is nice, but more useful would be the data in Postgres or Mysql. Unfortunately the queue module does not yet support ODBC DB logging (yet). In the meantime this quick and dirty hack gets the job done. Replace the flat file with a unix named pipe.

I've got a Zhone 24 port FXS to configure. The configuration is beyond stupid. The people that designed this unit should be chased down and fired. I'm going around in circles frigging with all the options. Does anyone have a config file for this unit that I can use as a starting point? -bill wlloyd@slap.net

Hie! I've installed Asterisk CVS-HEAD with ASTCC. The problem i'm facing is that the astcc.agi script completes when the recipient picks up the call. When the astcc.agi completes is returns 0 bill time but both end still able to talk. It occurs intermittently, any one facing the same issue? Asterisk Console ----------------- == Spawn extension (sip, 7777771111, 2) exited non-zero on

I'm trying to get a patch tested for inclusion in CVS. Anyone that is running E&M on a T1 and had to fool around with emdigitwait could you please try this. This patch removes the need for the emdigitwait parameter and speeds up dialing. This situation is mostly interfacing a legacy PBX/Key system with Asterisk. I've been running a couple of systems with this patch for

I'm sure this is easy and I'm missing something, but how can I add an extension into the directory command (ie get's it's list of extensions from voicemail.conf) yet have no voicemailbox for the extension? Basically I have an extension that gets forwarded onto somebody cellphone where they use their own voicemail. I'd like to be able to list them in the company

Hello, Here is my setup : Telephone 1 ( GXP 2000 ) Telephone 2 ( SPA942 ) Asterisk 1.4.17 ( same behaviour on Asterisk 1.4.23.1 ) Scenario: I don't like the default asterisk transfer feature, so I am trying to write my own. What I did : 1. Added to dynamic features #3 with AGI pointing to my php script 2. PHP script asks the user to enter his/her extension 3. PHP connects to Asterisk

Anyone know how (or if it's possible) to get a logo on the screen of a Polycom 501? I've been looking around for hints on how to do it but so far nothing would indicate Polycom supports doing it. -bill

I have looked into this callerid problem now for a few hours. 1) Caller id on a sipura-2000 now shows: cidname 2000 Where cidname is the new outputted formate from the cid_rewrite agi script and 2000 is the exten number. In looking at the Dial() application, option "o" 'o' -- Original (inbound) Caller*ID should be placed on the outbound leg of the call

Anyone having problems compiling the current cvs head this morning? New cvs checkout on RH9, followed by appropriate make clean and make install. System was running cvs head from Nov 23 with TDM card, PRI, SIP phones on local wire, and IAX. Appears zaptel and libpri compiled correctly, however the first attempt in the asterisk src directory yielded: gcc -pipe -Wall -Wstrict-prototypes

I updated to the latest CVS yesterday, from a version several months old. On one of my extensions, I have an AGI script in priority 1. Previously, the AGI script would run and when it terminated, asterisk would move on to priority 2 and connect the call. But now, when it terminates, it starts all over again in a continuous loop and never gets to priority 2. Do I need to update the priority in the

I have got call recording working on our 1.4.30 asterisk box together with a recording pause ability and being able to play different audio to each party at the start and end of the pause. This all works perfectly but one wish is to have the audio files have a beep or something in them so when you listen later you can tell where the audio was paused. So I changed things around so that instead

thanks Billy. I replaced print "STREAM FILE $filename \"\"\n"; with print "EXEC PLAYBACK $filename \n"; and it worked fine. Interestingly when I did print "STREAM FILE beep \"\"\n"; within the script, it worked. If I wasnt a newbie to asterisk I wouldve thought this to be strange. >From: "William Piper"

Emrah wrote: >> This is an asterisk-users question, and would have been more appropriate to have >> asked there. >> >> Instead of setting up your conferences in meetme.conf, you could set them up >> dynamically in the dialplan, and then you can control whether the user is >> prompted for a pin or not when joining the conference, based on whatever logic

Anyone having any problems with CVS-HEAD-12/24/04-15:59:15 and ALERT_INFO I have a system setup with polycom phones configured to auto answer on internal calls. When we upgraded to the latest CVS the auto answer stopped working. My dialplan has not changed. I did a sip debug and I dont see the alert-info tag in any of the sip traces. Any help would be appreciated. Thanks John Bittner Simlab.net

Greetings! I've searched far and wide for an answer and have gotten no where, so I was hoping one of you guys might have the answer; Is it possible to dynamically add a context to the dialplan? You can add extensions via the CLI, however if the context doesn't exist I get an error message instead of it creating the context for me. Any method will do, AGI, AMI, CLI... I just need a

This depends on your application. As you say you are able to do everything you require in dialplan at that is great. I have used AGI fairly extensively becuase the stuff I want to do can't be done in dialplan alone. For instance i have written a auto attendants that can be dynamically controlled by a non-techie user with real time and in call reconfiguration. Also i have written IVR apps that

Hi, I've been developing some CTI software around asterisk for a while, mainly with the help of AMI and fast AGI. It works quite fine, but I have some trouble sometimes with the un-synchronized property of these 2. Let me explain, we have a dialplan like this one : exten = s,n,UserEvent(useful_input_data) (...) a few actions exten = s,n,AGI(agi://127.0.0.1:3333/fetch,queuename) The idea is