similar to: Re : Re : Re : Communication IAX2 >SIP>IAX2

Dear list; I'm trying for forward some calls to an others asterisk using IAX2 protocol. But My asterisk can forward some calls and for others it hangs up automaticaly. Before my asterisk was working perfectly, i do not know what is happening!! When i try directly zoiper with my provider's asterisk it works perfectly. Here is the output from the cli when i made a call that asterisk hangs

I am looking for a way to Forward to a external or internal number and require a digit(s) in order to complete forward. Example: PSTN1 Calls * dials PSTN2 if PSTN2 presses proper digits bridge the PSTN1 and PSTN2 if no response return to a context Reasons: 2 actually 1. call is forwarded to cell phone but If cell is out of range, turned off,

Hello folks, I want to write an AGI script doing this: 1-user call a number. 2-asterisk call the agi script 3-the script dial the peer 4-if the call is answered, let the call up for 1min 5-then the script hangs up the channel. I tried either in php or in java but no success. In java i did this: ////////////// exec("Dial", "IAX2/400"); boolean t=true; while(t){

Hi, It seems that the "switch" parameter has a priority in the dialplan that is higher than the wildcard extensions.. This I am finding to be a problem.. My setup.. UA1--[AST1]--{IAX}--[AST2]--UA2 | | PSTN1 PSTN2 I use switch on AST1 to connect to AST2... As you can see I have PSTN connections on both and also the IAX connection is not permanent.. I

Sorry, If this is a dual post, was having trouble with email. I am looking for a way to Forward to a external or internal number and require a digit(s) in order to complete forward. Example: PSTN1 Calls * dials PSTN2 if PSTN2 presses proper digits bridge the PSTN1 and PSTN2 if no response return to a context Reasons: 2 actually 1. call is

Hello All! I am in the process of assembling an asterisk-based phone system for my office - using SPA-3000s to connect the network to the PSTN. I am wondering if anybody else can get (or has already seen) the same behaviour out of their 3000. The short version: Send multiple Calls to the SPA's FXO port at the same time it is re-registering with Asterisk. SPA HTTP Configuration:

Hi, I'm trying to bring up two PRIs from qwest with asterisk and dahdi. I'm using an OpenVox D410E and the drivers are loaded. My system.conf looks like this: # Span 1: TE4/0/1 "T4XXP (PCI) Card 0 Span 1" B8ZS/ESF RED span=1,2,0,esf,b8zs bchan=1-24 # Span 2: TE4/0/2 "T4XXP (PCI) Card 0 Span 2" (MASTER) B8ZS/ESF RED span=2,1,0,esf,b8zs bchan=25-47 dchan=48 These

List, Its been 2 weeks since my previous email and this time I am linking all 97 issues marked 'Ready for Testing' [1]. Simply follow the link, view the available patches, download, compile and install. Report your result into the actual issue, we can them continue to triage the issue. The more testers the better. If you have any problems or questions, jump on #asterisk-testing on

First I've tryed with the version 1.4.36 But it didn't worked so I supposed it should be ok with the last version 1.6.2... but not => I will create a new issue for this if you think it should be. Just hope it will not be too long to have a correction. Thanks a lot. Sebastien On Fri, Sep 24, 2010 at 9:11 AM, IMS <ims77.dev at gmail.com> wrote: > No ideas ? > Just give me

On Wed, Jan 28, 2015 at 12:23 PM, Ishfaq Malik <ish at pack-net.co.uk> wrote: > Hi > > We're using 1.8.23.1 on CentOS 5 and are trying to get accurate stats for > queues. > > For a particular customer, when I run queue show <queue_name> I get the > following numbers: > > <queue_name> has 0 calls (max unlimited) in 'ringall' strategy (17s

On Tue, Jan 27, 2015 at 4:14 PM, symack <symack at gmail.com> wrote: > Hello Everyone, > > I am required to write a java program that will get our asterisk to: > > * Query the database for phone numbers > * Loop through numbers and dial > * Play message > * Get dial pressed response > - If 1 = Yes > - If 2 = No > - If 3 = Connect to Agent

On Wed, Mar 4, 2015 at 12:47 AM, Jarrod Cuzens <jarrod at mogl.com> wrote: > For those that were interested I have attached the kamailio.cfg which we > have working with Kamailio 4.2.1 and Asterisk 1.8.23/32. Specifically, the > following yum packages: > > kamailio.x86_64 4.2.1-4.1 > @home_kamailio_v4.2.x-rpms > kamailio-auth-ephemeral.x86_64

On Thu, Mar 5, 2015 at 2:29 AM, Dmitry Melekhov <dm at belkam.com> wrote: > Hello! > > Just installed asterisk 13.2.0 and see many such messages in log, I see them > in console during calls, really something like this: > > > -- Executing [6166 at kanbaikal:2] Dial("OOH323/kanbaikal-6", > "SIP/6166 at asterisk") in new stack > == Using SIP

Todd How do you have the context in the phones sip configs set? Bryant From: "Todd Reese" treese65 at gmail.com Hi all, I've been have problems with getting this system on line and would like to acquire some help with the extensions.conf. My current problem is that the phones won't dialout.on the VOIP lines listed as dialout1, dialout2, dialout3. This version of asterisk

List, Over the last few months we have managed to bring the total number of issue on the tracker from 610+ to 537 (as of writing). While this is good news, we still have a number of open issues that require testers to help move them along. Below, I have posted the oldest 50 issues that are in the 'Ready for Testing' state. Basically, we are looking for more people to step-up and test

Hi, I have an odd problem when doing a blind transfer. The transfer is intiated and the transferred caller hears nothing until the timeout. I have tried setting the 'r' and the 'm' variables in the dial command. Nothing happens when I use the 'r' variable when I use the 'm' variable I briefly hear music on hold and then it stops until the timeout for no answer

Hi I am new to mailing list ,please correct me if the way of posting is not correct Relpying to : Re: make asterisk do something when an outgoing call is picked up (lee) For making asterisk do something on outgoing call Dial application is itself used Like for Playing an announcement to the caller on pick up the is an option A(x) where x is the file to play to the called party. Also

Hello All! I have a problem with my calls, that drops after 20 - 30 seconds. I got a piece of PAP2-NA log and Asterisk log and there's an error 603 - call declived, as showed. Thanks for any help. McCoy *********** PAP2-NA LOG *********** Feb 9 09:00:56 192.168.4.205 Feb 9 09:01:11 192.168.4.205 [0:5060]->192.168.3.14:5060 Feb 9 09:01:11 192.168.4.205 [0:5060]->192.168.3.14:5060

Hi there, I have a RoR app but for one of the funcationality (photo uploads) I am using PHP. I need to know how I can share/access session information from RoR in PHP since the PHP also updates the database and the entry information is contained in the RoR sessions. Please advice. Thanks in advance. Adil --~--~---------~--~----~------------~-------~--~----~ You received this message because

Hi, Did anyone here tried to use rsync between a unix machine and a NetApps Vfiler volume? I have strange case where it does create the folders on the vfiler volume but does not copy the files. Command is something like this: rsync -va dropzone pushacc@prod:/launchpad where, * dropzone is the the folder which keeps source files and folders in unix. This folder is owned by a unix account