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

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

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, 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.