similar to: Async Agi problem

Hi all, I have an external application commanding asterisk by AMI and AsyncAGI. I also have a dialplan like this: ; AsyncAGI extensions exten => _8.,1,Noop(entering in AGI loop at 8 ${EXTEN}); exten => _8.,n,AGI(agi:async); exten => _8.,n,Hangup(); ; Meetme extensions exten => _1.,1,Noop(Conference ${EXTEN} ${CONTEXT}); exten =>

Dear Folks, Anyone knows if Sangoma supports or going to provide support for battery removal detection on FXO lines?? As Tzafrir said earlier DAHDI supports it, which is a very nice feature but what about Sangoma? Regards. -- M. Shokuie Nia. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all, For a while now I've been using Asterisk together with HFC-PCI cards (Cologne chipset) for Euro-ISDN BRI support. However, I do not consider this to be the most reliable solution and believe that the most stubborn problems have always been software related. If my clients are willing to spend a bit more money on different hardware, what do you think the best solution would be?

Howdy, I've spent a couple of days writing a new feature for Asterisk that allows to record calls in T1 or E1 PRI lines using Asterisk connected to tapped lines. This means that you don't have to install anything in the PBX's/telco equipment that is going to be monitored, all you need is to install a device like the PN 633 Tap Connection Adapter that is available for example, from

I'm converting Ubuntu startup scripts to work on CentOS (5.3), and I'm having trouble finding out how to start a daemon in a certain directory? For Ubuntu, start-stop-daemon has the option -d to set the working directory. TIA /Rob

Hi all, I've setup two * servers which are SIP interconnected ala osaka/toronto from the * book (before anyone sugggests using IAX instead, no, I NEED to have them SIP interconnected for verification/test purposes). Then I have a Zoiper connected to one of them via IAX (so that * will not reinvite (?)). As soon as I try to call (via Zoiper) an extension on the other * I get a "Failed to

I have been trying out several stun servers with Asterisk 1.6.0.9 and 1.6.1.0 and I keep getting the following message: [May 26 12:26:35] WARNING[16174]: chan_sip.c:2695 ast_sip_ouraddrfor: stun failed [May 26 12:26:35] WARNING[16174]: chan_sip.c:2695 ast_sip_ouraddrfor: stun failed [May 26 12:26:35] WARNING[16174]: chan_sip.c:2695 ast_sip_ouraddrfor: stun failed [May 26 12:26:35] WARNING[16174]:

Hello everybody, I need help, I have a problem with conferences in asterisk, when many people are in a conference sometimes there're users pressing phone keys and this action emits a sound (DTMF of the phone keys), so, I need to find the way of not listening this sound.. I'm using MeetMe(variable,pFX).. I tried whithout "F" but it doesn't work because users continue

Hi All, I've been lurking here for a while now, having only made a couple of posts. I am starting a new hardphone project and was wondering if there is some GPL'ed IAX source that I could start with. I've searched and haven't come up with much beyond iaxClient. While iaxClient does give me a little bit to start with, it looks like it is really intended to be more of a

Hello Everyone, I've successfully registered my g729a licenses. When i try to load the module from asterisk Cli i got the following error *Error loading module 'codec_g729a.so': /usr/lib/asterisk/modules/codec_g729a.so: cannot restore segment prot after reloc: Permission denied* * loader.c:795 load_resource: Module 'codec_g729a.so' could not be

Hi I was reading this article on installing asterisk 1.6 + debian http://www.howtoforge.com/installing-and-configuring-asterisk-1.6-and-postgresql-to-manage-cdr-and-realtime-config-on-debian and I noticed they suggested to recompile to 1000Hz enable kernel, I currently have a 250Hz stock standard kernel. I am running on a soekris board - amd geode cpu. Is recompiling the kernel to the

Is this patch correct? The "&&" doesn't make logical sense to me. I think it should be "||" and making this change fixes the problem I have with SIP phones in MeetMe conferences. If it's correct, is there someplace more formal that I should submit it to? *** app_meetme.c.old 2009-10-11 17:56:44.000000000 -0400 --- app_meetme.c 2009-10-17

Hi, Has anyone ever experienced crosstalk with sangoma analogue cards on 64bit? We have exhausted every test to try and replicate this and find a solution with Sangoma tech support, but we can not fix it. We are about to try the card and four *seperate* UK BT lines in a 32bit system. The current system is a 4gb, dual core cpu with pbx in a flash 1.4, Zaptel and Asterisk 1.4.21-2 Currently we

I'm dealing with an idea to exchange data in a socket connection style or a sort of ftp transfer with IAX2 as the transport medium. An IAX client on e.g. a notebook could establish a connection to any remote machine (also client) via any Asterisk Server where both clients are registered. Due to the unique properties of IAX2 one could connect quite easily to any "hidden" remote

Hello, I have previously open a topic on the mailing list about deadlocking on Asterisk 1.2.35. After upgrading to 1.4.29.1 we still experienced the same problem : Mar 5 12:05:56] DEBUG[8647] channel.c: Avoiding initial deadlock for channel '0xb7689840' [Mar 5 12:06:41] DEBUG[7130] channel.c: Avoiding deadlock for channel '0xb7c04788' [Mar 5 12:06:41] DEBUG[7130]

Hi, I am developing asterisk support for our application using the Async AGI and Asterisk-Java. One thing I haven't been able to implement is how to stop playing a sound. Something similar to StopIO for Dialogic GlobalCall or DivaStopSending for Eicon. Is there any way to achieve this today which I have missed? Or could someone give me hints on how I could implement this in the res_agi.c The

Hi, Is there a way to execute next priority in the dialplan if you have called agi:async? I want to play warning message if adhearsion is down. Currently I wasn't able to make it work. The dialplan execution ends after the first priority. [incomming] exten => _X.,1,AGI(agi:async) exten => _X.,2,Answer exten => _X.,3,Playback(some-message) exten => _X.,4,Hangup Regards, Pavel

Hello, i'm using asterisk 1.4.9 in fedora 7, i was compiled its with this package: libpri-1.4.2 asterisk-1.4.9 spandsp-0.0.4 unicall-0.0.5pre1 libmfcr2-0.0.3 libsupertone-0.0.2 libunicall-0.0.3 zaptel-1.4.4 i'm using a E1 pci card with R2 but they not work, when I start the asterisk its generate this log: [May 12 08:53:24] WARNING[30814] channel.c: No channel type

I have a 'CONGESTION' Status with R2 protocol. While testing this scenario sip GW--?Asterisk ?Digium E1 R2 Protocol?Cisco E1 R2 protocol?sip Gw Find below my error and configuration ,where are the errors in my configuration ? ========================================================================= Connected to Asterisk SVN-branch-1.6.2-r235775 currently running on

It means, AMI application is no more running or crashed or lost network connection with asterisk server. In such cases call is neither answered nor disconnected by Asterisk. I want to detect such state and jump to next dial plan to answer or reject the calls Regards Amit Patkar On September 20, 2016 8:07:23 PM GMT+05:30, Matthew Jordan <mjordan at digium.com> wrote: >On Sat, Sep 17,