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Asterisk Users Mailing List - Non-Commercial Discussion
December 2009 - page 1
AGI (1 Reply)
Thomas Perron — 125963259901 Dec 2009*
I am trying to find an AGI script that runs via PHP and performs the send text application. Does anyone have any tools or scripts set up for this plea...
Polycom retrieve call from hold (13 Replies)
Mike Diehl — 125963760501 Dec 2009*
Hi all, I've got a Polycom 501 that's been working with Asterisk for some time. However, I don't seem to be able to put a call on hold and...
Rizwan Hasnani — 125965327401 Dec 2009*
hi, i am using asterisk 1.6 and i want to integrate sphinx speech engine with my asterisk, so that i can use the generic speech API provided by asteri...
toqeer ali — 125965571501 Dec 2009
I am trying to configure Freeradius with asterisk.what i am doing is 1) install Freeradius 2) Radius Client confiugration is ok in "server...
Issue with T38 fax Calls (2 Replies)
Venkatesh Arun-A21466 — 125967208001 Dec 2009*
Hi all.. Im using ABE C3.2.1 version and here Im having the following issue with respect to T38 fax calls. Somehow asterisk when it receives Session m...
Magnus Benngård — 125967232801 Dec 2009
Gentlemen, Forgive me if I am posting at the wrong place! I was going to test the "new" chan_ooh323 driver so I did install: debian: Linux s...
Leif Neland — 125967918801 Dec 2009*
I made a patch to app_dial.c to make Dial(ext1&ext2&ext3,tumeout,B) return busy when just one extension is busy. http://www.neland.dk/app_dial...
Asterisk registers with private IP (5 Replies)
Joao Gomes Pereira — 125969628601 Dec 2009*
Hello I'm trying to register an Asterisk working behind Nat. Here is the trunk: register=username:password at sip.startel.pt [startel] type=peer h...
zac wolfe — 125970058401 Dec 2009
SafiServer and SafiWorkshop 1.2 is here! This is a seminal release for us as the product is now more stable, powerful, and easy to use than ever. We...
Ignacio — 125970455401 Dec 2009
Hello everyone, I am trying Asterisk could manage codecs negotiations. I have some telephones that supports g723.1 and G711, while others only support...
"Dropping incompatible voice frame" error (2 Replies)
Richard Kenner — 125970912001 Dec 2009*
I have a SIP phone calling an AGI application. It starts out this way: -- Executing [s at macro-Call-AGI:2] AGI("SIP/151-b414f0c8", ...
OpenSBC (2 Replies)
gergis.rasmy — 125971035201 Dec 2009*
does anyone use OpenSBC , or know if it is mature stable opensource for a production enviroment http://rpm.pbone.net/index.php3/stat/4/idpl/1......
Question about g729 (5 Replies)
Landy — 125973924802 Dec 2009*
Hello. I am currently testing an asterisk server using the default codecs, I have allow=all, and noticed everytime I test it in a wireless lan the lat...
Dan Journo — 125974717602 Dec 2009
Sorry for the repetition. I didn't see the other responses. -Original Message- From: Dan Journo Sent: 02 December 2009 07:36 To: Asterisk Users Ma...
Featuremap help (1 Reply)
Doug Crompton — 125975056402 Dec 2009*
Using version 1.2.35 built by root @ slate on a i686 running Linux on 2009-09-15 00:24:10 UTC Problem - I cannot get featuremap right. Have added a fe...
Question about g729 (2 Replies)
Dan Journo — 125975938202 Dec 2009*
However, I've read somewhere that passthrough doesnt require a license. Which means that if your sip clients can transmit in g729 and your voip pr...
d tbsky — 125976591102 Dec 2009
hi: our country use ETSI Standard ETS 300 659-1 to send caller-id. the caller-id format may be DTMF or FSK. with the newest patch in issue 9096 (9096-...
Slightly OT - Oreka Call Recording (5 Replies)
Tim Nelson — 125976824002 Dec 2009*
Greetings all- I'd like to install Oreka on a Centos 5.x server for monitoring my Asterisk systems(using port mirroring) but I find I'm having...
Matt King — 125977223602 Dec 2009
Hello, Just to let you know, our popular tutorial on setting up Asterisk for call centres has been updated. The tutorial covers everything from initia...
Eckhard Jokisch — 125977676302 Dec 2009
Hi, I am stuck with my analog telephones on DADHI/1. I can place a call from a SIP-phone to the analog phone. But as soon as I try to dial somethin on...
Asterisk Development Team — 125977835802 Dec 2009
The Asterisk Development Team has announced several Asterisk-Addons releases, including Asterisk-Addons 1.4.10, 1.6.0.4, and 1.6.1.2. These releases a...
b option in Directory (5 Replies)
Martin Roy — 125978747302 Dec 2009*
I'm running asterisk 1.4.21 and I see if I go on the wiki that there's a b option that let you enter the first name OR last name of a user. I ...
James A. Shigley — 125978822202 Dec 2009
It might be worth mentioning the voip call is coming from a number we have thru bandwidth.com in case anyone uses them. James Shigley From: James A. S...
Glen Ganderton — 125979182202 Dec 2009
Hey Guys, I have installed AsteriskNOW and I've found that the SIP module is not installed (along with all the SIP configuration files) until I lo...
Problem with Timeout (3 Replies)
Dan Journo — 125979197202 Dec 2009*
Hi, I have a problem with incoming calls. They all seem to be ending after 600 seconds (10 minutes). I've added:- exten => _X.,2,Set(TIMEOUT(ab...
Variable Name needed (4 Replies)
James A. Shigley — 125979304502 Dec 2009*
Other than having stripping out IPs this is what I am receiving for my voip calls. Now I normally use ${CALLERID(rdnis) for RDNIS, this works fine cal...
Doug Crompton — 125982306003 Dec 2009
Using version 1.2.35 built by root @ slate on a i686 running Linux on 2009-09-15 00:24:10 UTC Problem - I cannot get featuremap right. Have added a fe...
Flashing Cisco 7941 to SIP (1 Reply)
Ricardo Melendez — 125982341803 Dec 2009*
Hi to All, I am trying to flash to SIP image one Cisco 7941 IP Phone to work with Asterisk, I have searched the internet and find some instructions bu...
Remco Barendse — 125982525103 Dec 2009
I never do sip show channels but i tried it this morning to see if everything is working after upgrading 2 boxes to 1.6.0.19 and 1.4.27.1 Is it correc...
Hans Witvliet — 125982847303 Dec 2009
Perhaps slightly O.T. Does anybody know of (or even better, has experience with) softphone clients on a blackberry? Some friends of mine have those de...
Roger Schreiter — 125983536803 Dec 2009*
Hi, I'm using Sangoma's wanpipe together with dahdi, all software downloaded today at most recent version. Hardware is Sangoma A104, a 4xE1 ca...
Dial application with M option (2 Replies)
ABBAS SHAKEEL — 125983667703 Dec 2009*
Hello, What i am trying to do is ..... Dail a number and ask if you wana talk to XXX press 1 and if you dont wana talk press any other key. For this p...
Olivier — 125983859303 Dec 2009
Hello, How can you parse a comma separated list using function CUT and AEL ? I've tried but it displays error message (though is seems to find the...
Olivier — 125983894503 Dec 2009
Sorry for the noise but I mixed up with another mistake elsewhere in my code. The above example works !...
Joao Gomes Pereira — 125984530203 Dec 2009
Hello I have a simple configuration to allow the admins listen to agent calls: exten => _654,1,ChanSpy(Agent) exten => _654,2,Hangup() The probl...
Wi-Fi sip phones with auto provisioning (4 Replies)
Lefteris Zafiris — 125986574703 Dec 2009*
Im looking for wifi sip phones that support auto provisioning and work flawlessly with atserisk. Can anyone suggest me some models?...
Philip A. Prindeville — 125986955203 Dec 2009
I've recently decided to spend idle cycles while waiting for various Astlinux platform builds to complete on making the contents of asterisk/confi...
Olivier — 125988241003 Dec 2009
Hello, Has someone successfully used this QUEUE_VARIABLES() function (in 1.6.2-rc7) ? I tried to use it as I'm using SIPPEER() but without success...
Feature Request: "SayNumberFiles" (4 Replies)
Olivier — 125988271803 Dec 2009*
Hi, Currently, it seems impossible to use the output of SayNumber application as an input to Read application. So, if you want to develop an IVR with ...
Source-IP on Asterisk DRBD/-HA-Cluster wrong (8 Replies)
Thorolf Godawa — 125989258404 Dec 2009*
Hi all, I installed a Linux-HA-cluster with DRBD and Asterisk 1.4 on it. Actually it might work quite good, failover etc. works, even if this is not a...
Multiple Channel Variables with AMI Originate (3 Replies)
Matthew Edmondson — 125989920704 Dec 2009*
Hi guys I seem to be having a problem, I don't know if it's a bug or whether I'm just doing it incorrectly. I want to set about 3 channel ...
DHAVAL INDRODIYA — 125990617204 Dec 2009
hello all, i found this error on asterisk CLI while try to play file on SIP channel. scenario is, manager send a command in meetme and meetme will pla...
Masood Ahmed — 125991177504 Dec 2009*
hy Hope everyone is fine, I have one issue coming in asterisk , What i am doing is i am generating a callback if some one calls at a specif access num...
Masood Ahmed — 125992026304 Dec 2009*
Dear Hakan, thank you for your information on this issue it does change but it only changed in REQUEST URI field not in From Field, Date: Fri, 4 Dec 2...
Leif Neland — 125993613004 Dec 2009
I'd like to put a phone in a special context, where a test is made on its business hours, then if so, proceed to the normal context to do whatever...
Randy R — 125994402304 Dec 2009
VoIP Users Conference begins in about 30 minutes to discuss the use of VoIP on social networks like Facebook. If you have any interest in this (or may...
Olivier — 125994514504 Dec 2009*
Hi, I'm using revision 6822 of Dahdi Tools. # dahdi_hardware pci:0000:05:06.0 wcb4xxp+ d161:b410 Digium Wildcard B410P # asterisk -rx "dahdi ...
DAHDI issues on 1.4.26.1 (1 Reply)
Mike — 125994559204 Dec 2009*
Hi, Running 1.4.26.1 here. I have installed TE420B card in my server, and followed the appropriate steps (as far as I know to configure it). This TE42...
Olivier — 125994657504 Dec 2009*
How can can you get current queue's length (ie maxlen) or waiting call number from dialplan ?...
James A. Shigley — 125994774304 Dec 2009*
Trying to configure IAX for use I think I have everything set right. But my IAX phone wont connect. When I run wireshark I'm seeing this Note if a...