similar to: AGI command

Hi all, I just started programming using AGI and I have a simple question about STDERR. If I understood it right, all the messages sent to "STDERR" should be shown in the Asterisk console, but using the following python code I just can't see anything. #!/usr/bin/python # # File: /var/lig/asterisk/agi-bin/agi-test.py # # Description: An AGI Script # import sys env =

Hi folks, A user here has asked if we can display the current voicemail message's envelope (date/time/caller id of message) in the LCD of the Polycom phones we use (430 & 501). I realize this is somewhat like the many caller-id-after-the-fact threads, but I figured maybe someone had solved this a different way. Has anyone been able to do this, via caller ID, messaging, the mini-browser

Hi, I'm trying to find a way of passing the actual call duration (something like ANSWEREDTIME) to an AGI script that runs periodically during a call. Any ideas? Thanks, Adi. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070531/115b0aee/attachment.htm

I am working on a call-back solution where the initiating call should never be answered. I was doing this simply through the dial plan, sending a progress tone, and then dumping the channel, and firing off a DeadAGI which created a call file to make the callback. Now I've tried extending this so that an AGI is fired first to check for things - like no inbound ANI - and play a

Hi all, I just want to know how I can make sure that incoming calls to my asterisk server are being treated by [incoming] section of extension.conf file. Thanks in advance. Ronaldo.

Hi all, I have a IAX trunk between two asterisk servers, both with dynamic IP and both have a DNS name associated with it. In the iax.conf file I configure the "host" parameter with the DNS name of the servers. Everything works fine until one of these servers get a new IP, so the other can't find its peer (the one that has just gotten a new IP). If I manually issue a "iax2

Hey folks, So I'm trying to get Festival() working on 1.2.17. I'm trying to use app_festival: Here's the show dialplan output from that extension: '3378' => 1. Answer() [pbx_config] 2. Festival(Hello Asterisk caller. How is your day?) [pbx_config] 3. Playback(vm-goodbye) [pbx_config] 4. Hangup()

Hi all, What does (T) mean on the output of "iax2 show peers"? The following my output. darkstar*CLI> iax2 show peers Name/Username Host Mask Port Status ronaldo (Unspecified) (D) 255.255.255.255 0 UNKNOWN sp/ata 201.26.67.102 (S) 255.255.255.255 4569 (T) UNKNOWN 2 iax2 peers [0

All, I've a question on A*k hardware. I'm running 1.2.18 on a Dell DC051 (Intel(R) Celeron(R) CPU 2.80GHz) with 512mb RAM. I'm supporting 60 users (Cisco 7940s each + Xlite PCs). Call loads are low, max of about 10 simultaneous SIP/IAX calls. CPU for A*k rarely goes above 2% as I can tell. Its IP only, no E1/T1 cards present. However, I get complaints of bad voice quality,

The bar is getting raised yet again http://www.voipmonitor.net/2007/06/05/New+Features+And+Services+For+Pack et8+Virtual+Office.aspx I personally use Snapanumber $30 or there abouts (after trialing a few other TAPI solutions and finding them sub-par) and think it's a great product but interesting to see how more people are expecting desktop/phone integration applications. Does anyone

Hi, I''m trying to configure QoS on my linux in the following manner: I have a main link with 64K, so I divided it in 3 classes of 18K, 14K and 9K with an excess (not used for classified traffic, just to be shared) of 23K. This excess should be distributed proportonally among the 3 classes, that is, the class that has more rate should borrow more bandwidth. What is happening is

Hi, I just got a Polycom 330 and, of course, I don't have the firmware and sip.cfg files to provision it. Where can I get those? Mike -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070605/a14615c8/attachment.htm

Dear Users ! I have recently installed asterisk 1.4 i got a warning message whenever i use reload or extensions reload. [Jun 29 19:22:11] WARNING[4539]: pbx.c:6236 ast_context_verify_includes: Context 'ael-local' tries includes nonexistent context 'ael-parkedcalls' [Jun 29 19:22:11] WARNING[4539]: pbx.c:6236 ast_context_verify_includes: Context 'ael-dundi-e164-local'

I am trying to get updateconfig working. I found an example of updating configuration files here: http://www.voip-info.org/wiki/index.php?page=Asterisk+Manager+API+Action+Upd ateConfig When I tried it the conf file was updated but the new entry was not added. action:updateconfig reload:no srcfilename:manager.conf dstfilename:manager.conf Action-000000:append Cat-000000:newuser

Hi all, Does anybody know any USB phone that I can use as an IAX Client? Thanks. Ronaldo.

Hi , > > My Xen has been working fine for a few days. Then today it suddenly > can not run. > > When it runs It give me this error : - Error starting domain: virDomainCreate() failed POST operation failed: (xend.err "Error creating domain: Boot loader didn''t return any data!") Details: - Traceback (most recent call last): File

I'd like to be able to generate a Manager Event from the dialplan but can't seem to find a way to do it. Alternatively, trigger and Event when a record in AstDB gets changed. Can anyone point me in the right direction? Thanks. By way of explanation, I've a app that connects to astmanproxy and I'd like it to know when a call group gets put into Nightservice. Putting the call

Hello there, I appreciate any help about this problem that I can't figure out... I need to record all my calls: this is pretty easy using Monitor() before the Dial(). eg: exten => 425,n,Monitor(wav49,/var/spool/asterisk/monitor/425/${EPOCH}_${CALLERID(num)}_in,mb) exten => 425,n,Dial(${PHONE1},10) Now, I want to create a call group: I mean, I want a number (eg 800) that makes

Hi all, I'm planning to deploy many Asterisk servers for remote sites connected through IAX. Behind each server, there will be many sip clients connected. A sip client from one site must be able to make calls for the other sip clients connected to the other remote Asterisk servers. I've heard that DUNDi is a good option in order for each Asterisk server to locate the right (or the

Hi all, Can some one suggest me an IAX client for Linux and Windows? I used KIAX once, but know it seems complicated to have it working on Ubuntu. Thanks. Ronaldo.