similar to: console debugging

Dear, I have asterisk 1.6.2.13 in a small hardware PBX (512 GB RAM and Celeron), and last days when I call from one extension to another of the same PBX after I dial the number the rings sound after 20 seconds. In the CLI log, when I debug the AGI, I see always goes good until dialparties.agi, and after that there are 20 seconds without any log, and so the ring sound. I've read

Hello: We have a TDM2433E Digium Card (12 FXS, 12 FXO) and Asterisk 1.8.7.0 running. Everything seems to be ok but call transfers. This is the issue: *A, B, C and D are in FXS ports*. 1) A calls B. B anwers. 2) B tries to transfer the call to C dialing *2 (code for attended transfer). 3) A hears MOH. B dials number C. 4) Asterisk says the dialed number is incorrect or non existing. We tried

Hi, I've been meaning to fix my non-working paging feature here for a while, and I've just spent the last 5 hours looking at many, many web pages that all say the same thing. I am using Asterisk 1.6.2.18 and Polycom phones, both older (501 with "latest" legacy 3.1.7 firmware) and newer (335 and 650 with latest 3.3.1f). I have changed the correct values in sip.cfg like

I'm trying to decide if I need to open an issue for this or if it's just a misconfiguration issue of some sort. Here's the situation - yesterday morning, I downloaded asterisk 1.8.19.1 and installed it on a fresh CentOS 5.8 installation and got a shell of a basic asterisk install setup (minimum required configuration files, etc, with no dialplan or sip peers setup yet). In the

Hi All; When the call coming via the E1 dahdi and I handle the call (as first step) by exten => 5631040,1,Goto(OrangeCMG,s,1) it is not working and the call will be disconnected instead of queued. But, when I handle the call (as first step) by playing any sound file and then send for the queue, then it is working fine, WHY? exten => 5631040,1,Playback(WelcomeMessage) exten =>

Hi All; If I set a context other than the default context, then I do not see a generation for a configuration in the extensions_additional.conf for this context, but always the generation for the configuration is for the default context (from-internal). Normally, I have to put some Phones in a context and another Phones in a context, and give each context a privilages, but if I do this, then I

Hey Guys! I am getting this wired error when i am calling IAX trunk. Everything works! but i want to get rid on these RED WARNING messages.. what is wrong here ? I have func_aes.so module loaded. also i remove and test but still same error. -Satish == Using SIP RTP CoS mark 5 -- Executing [7623 at from-sip:1] Macro("SIP/7527-0000000d", "orasebcamdial,7623") in

Hi All; Again, the Cisco IP Phones 7942G and using Skinny: I upgraded the firmware to version 8.5 (skinny) and I am using skinny channel (chan_skinny) and the skinny.conf file. The phones are registering, but when we use them to place a call, we only hear tooooooooooo in the handset and we do not hear voice (even when we dial the digits, we only hear toooooooo .. but it dials and destination

Hello, For those customers with only analog lines, who ask for fax2email and email2fax, whats the most reliable solution available and tested with Asterisk? Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120529/3e28b56e/attachment.htm>

Can you say a Virtualized Asterisk with a PRI card! http://www.phoronix.com/scan.php?page=news_item&px=OTY0OQ Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety."

I am using Elastix. Asterisk is used for PBX application in Elastix. I want to make customize MOH. But not able to use MOH. I make simple extension in asterisk conf file but no success :( Below are the details of configuration files. Even default MOH is also not working.... *Asterisk Version 1.6.2.17.2 * *1) Extension.conf* [incoming] exten => 6000,1,Answer exten =>

When I receive a call, I want to automatically park it from the dialplan so that I can retrieve it later. However, I don't want callers to be aware that they are being parked, so I want to play a ringing tone to the caller. Park() is supposed to be able to do this: Park([timeout][,return_context[,return_exten[,return_priority[,options[,parking_lot_name]]]]]) options r: Send ringing

I am running Asterisk 1.8.10.1. I am trying to set up some routing in my dial plans and having some issues (the issue being that I don't quite understand all of the syntax and patterns that can be used: Examples: DID1 = 6140000000 DID2 = 6140000001 CNAME1 = 6149999999 CNAME2 = 6149999998 CNAME3 = 6149999997 context1 context2 context3 I have looked at several examples (patterns) and I

Hi all, I'm getting one way audio when calling over the SIP trunk i.e. end device B (remote end of SIP trunk) can hear device A (softphone registered with Asterisk) but device A can't hear device B. Even though I configured same NAT configurations on other servers and they are working good. The NAT configuration is listed below; localnet=130.0.0.0/130.0.0.0 externhost=12.131.12.13

someone have been installed Asterisk (Trixbox) on VirtualBox which is installed on a Linux host (Ubuntu server 10.04 specifically). I want to know if it is convenient or not, and the reaseons if i should on shouldn't do it. Thanks in advance.! -- Esteban L. Cacavelos de Amoriza Cel: 0981 220 429 -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all I'm trying to configure my Asterisk setup to load the musiconhold.conf file from an ODBC connection to MySQL, working through the example given in the excellent book "Asterisk: The Definite Guide". I'm using Asterisk 1.4.19 and MySQL 5.1.58. I've configured the ODBC bit and in my GeneralLog on MySQL I can see the asterisk user connecting and sending a few SQL

Hi All; Using the FreePBX, after I added the extension from the GUI, I discover that it is automatically added in the extensions_additional.conf in the context [ext-local] and [from-did-direct-ivr] How I can change these context name? I need to determine this. How? Regards Bilal

Is there a way for asterisk's voicemail to send an email (including voicemail attachment) to multiple email addresses? -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110408/8908db5f/attachment.htm>

Hello, We want the ability to choose from an AGI script whether or not to enable faxdetect for calls over SIP or DAHDI. Is this possible, or can anyone suggest a workaround? Thanks for any advice. -- David Cunningham, Voisonics http://voisonics.com/ USA: +1 213 221 1092 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 8063 9019 -------------- next part -------------- An HTML attachment was

Hi ALL, still having trouble getting Ast 12 to run. I got it compiled and built but now when I try to run, I'm getting a missing library error that seems to be in error (see below). The .so file DOES exist with correct permissions. [root at Asterisk12 ~]# asterisk -rvvv asterisk: error while loading shared libraries: libasteriskssl.so.1: cannot open shared object file: No such file or