similar to: From external IP am not able hear the audio on the SIP extensions

Hi, i am trying to install asterisk newer version on the Elastix machine, but while installing the chan_sip,c module is not building while make. when i see in make menuselect options it showing "XXX" -- extended , please let me know how to enable it and make build chan_sip module. -- Upendra -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi ALL, Am new to Elastix and wanted to try build new modules in the Elastix , so i want to know how the PHP is running ?? as i see no Apache server inside ?? so wanted to know how its running ? which server and architecture? *--* *Thanks & Regards* *Upendra* -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, i am using elastix 2.3 and created some dahdi extensions,now i dialing between the extensions i.e like 2000 to 2001 , but there is delay of 3 to 4 second before it ring the destination. so cany anyone know how fix it so that after dialing the digits the destination should ring . without any delay after dialing. regards Upendra. -------------- next part -------------- An HTML attachment was

Hi, i wanted to know that if i have a message indicator SIP phone , then MWI will work in ELASTIX ?? Let me know the Details of MWI and how test it. *--* *Thanks & Regards* *upendra* -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20140804/125551b7/attachment.html>

Hi, I am trying to write dial plan for sip to auto answer (auto attend) the incoming call to the sip phone. - If i call from sip1 to sip2 then sip2 should automatically answer the call and play some sound file. I am trying to do this but as new to the asterisk dial plan configuration , so not able Todo this. help me if anyone already done this setup. Regards Upendra. -------------- next part

Hi, can anyone help me how to setup a simple gateway for voip phones on elastix system. I dnt no really how it should be connected in reality...? and how to test it . Regards Upendra. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20121210/1c9556eb/attachment.htm>

Hi I'm new to this list, so please forgive me off-topic or RTFM-questions. I have an asterisk/elastix driven phone-environment using Polycom SoundPoint IP 650 as extensions. When adding just one custom ringtone (~57KB) in a proper format (ML.wav: RIFF (little-endian) data, WAVE audio, ITU G.711 mu-law, mono 8000 Hz) the phone boots well. But after I have chosen the custom ringtone as my

*Dear List* Plz help, i am not much experienced with asterisk. i configured it on ubuntu 12.04. no problem when i call any mobile no(0091XXXXXXXXXX) but when i call on my local asterisk no.(101,102 or 105) it is not connecting giving error "Auto fallthrough, channel 'SIP/lucknow-0000006f' status is 'CHANUNAVAIL' *while when i call 200 it is playing audiofile successfully.

Hi all! I have trouble with connection to AMI 1.1 wich enabled on Elastix "*Asterisk Call Manager/1.1* *Action: Login Username: admin Secret: qweasd123* *Response: Error* *Message: Missing action in request*" Elastix versions: "* Kernel* * Linux(x86_64)-2.6.18-348.1.1.el5* * Elastix* * elastix-2.4.0-1* * elastix-portknock-0.0.1-0* * elastix-agenda-2.4.0-1* *

We have a problem in that calls are dropped after 15 minutes (on both internal and out going calls, incoming calls do not seem to have that limit) How do we fix it? This is the version on that PBX Kernel Linux(x86_64)-2.6.18-371.1.2.el5 Elastix elastix-2.4.0-8 elastix-a2billing-1.9.4-5 elastix-addons-2.4.0-10 elastix-agenda-2.4.0-14 elastix-asterisk-sounds-1.2.3-1

FWIW the tcpdump will be a few days at http://rosa.stappers.nl/ff/tcp_w.pcap md5sum a8090ca85202d62ddd5a3d5684b2e8fd tcp_w.pcap ----- Forwarded message from syslinux-owner at zytor.com ----- Date: Thu, 06 Jun 2013 12:49:11 -0700 From: syslinux-owner at zytor.com To: syslinux-owner at zytor.com Subject: Syslinux post from upendra.gandhi at gmail.com requires approval Message-ID:

I On Wed, Jun 5, 2013 at 7:41 PM, upen <upendra.gandhi at gmail.com> wrote: > > > > On Wed, Jun 5, 2013 at 7:34 PM, upen <upendra.gandhi at gmail.com> wrote: > >> >> >> >> On Wed, Jun 5, 2013 at 7:25 PM, Gene Cumm <gene.cumm at gmail.com> wrote: >> >>> On Wed, Jun 5, 2013 at 7:49 PM, upen <upendra.gandhi at gmail.com>

Hi, i am trying to install the Asterisk test suite on my ubuntu system , i have followed all the installlation steps as mentioned in the link ( http://blogs.asterisk.org/2010/04/29/installing-the-asterisk-test-suite/) , but when i am trying to run the script some of the test cases are PASSED and most of them are FAILED and SKIPPPED. So please help me out to do the testing correctly. The

On Wed, Jun 5, 2013 at 7:34 PM, upen <upendra.gandhi at gmail.com> wrote: > > > > On Wed, Jun 5, 2013 at 7:25 PM, Gene Cumm <gene.cumm at gmail.com> wrote: > >> On Wed, Jun 5, 2013 at 7:49 PM, upen <upendra.gandhi at gmail.com> wrote: >> > On Wed, Jun 5, 2013 at 5:39 PM, Gene Cumm <gene.cumm at gmail.com> wrote: >> >> On Jun 5,

This Wednesday, November 26th, the Toronto Asterisk Users Group invites all in the area to join us for a telephony workshop and talk sponsored by Sangoma Inc.[1] Jose Landivar, co-founder of PaloSanto Solutions[2], creators of Elastix, will be running a "getting started" workshop on Elastix, followed by a talk discussing how it differs from other Asterisk-based distributions, and a

Dear community, Please help. I've been looking around the internet (and in this great forum) for help with DUNDi setup between servers (I'm using Elastix) and while I can get my servers to lookup extensions on each other very well, I have not been able to successfully make calls between servers. For my test environment, I have 3 servers setup for now, and these are the steps I've

On Wed, Jun 5, 2013 at 7:49 PM, upen <upendra.gandhi at gmail.com> wrote: > On Wed, Jun 5, 2013 at 5:39 PM, Gene Cumm <gene.cumm at gmail.com> wrote: >> On Jun 5, 2013 6:13 PM, "upen" <upendra.gandhi at gmail.com> wrote: >> > >> > Thanks hpa. It looks like it's not able to find lpxelinux.0 files which >> > actually resides under

---------- Forwarded message ---------- From: Juan Carlos Huerta <juancarloshuerta at gmail.com> Date: 07-abr-2009 13:41 Subject: Re: [asterisk-r2] MFCR2 With Unicall - can unicall affect zaptel commands? To: asterisk-r2 at lists.digium.com Please wirte to asterisk-users at lists.digium.com to get help about this problem. Juan Carlos ~ Lo que no te mata te fortalece ~ On Tue, Apr 7,

I am using Polycom 501s with asterisk 1.2.4. When transfering to call parking wih "#1" -> 700 the user is able to hear asterisk tell him what extension the call was parked on. However, when I press "transfer" -> blind -> 700 . The user is not able to hear what extension the call was parked on. It seems like the polycom is hanging up before asterisk is able to

Hi; I used vicidial for call center and I would like to try elastix. Can someone advise about the advantages? Does Elastix has a screen for the agent to login/logout from their PC and deal with the inbound/outbound calls and Integrated with the CRM? Regards Bilal