similar to: Caller ID : FSK ETSI or FSK US

Hello, Currently my asterisk system pickup incoming call after 3 or 4 rings. How can I ask it to answer the call on the first ring? I put immediate=yes on /etc/asterisk/chan_dahdi.conf but result in no different. Thanks in advance :) BR, Anam -- Sent from my mobile device

In order solve my incoming caller ID problem, I upgrade the dahdi to version 2.6.1 from version 2.4.x. After upgrade, I found the echo cancellation doesn't working (I'm using Digium AEX800B PCI Express card). I can hear my self talking on the phone. How to solve this? I think I need to recompile dahdi 2.6.1 with OSLEC support? how? [root at callcenter ~]# dahdi_cfg -vvv DAHDI Tools

All, has anyone any experience in using Wifi smartphones as SIP clients? Does this work properly? What models/brands are optimal for this (in terms of ease of use, battery life etc)? Thx!! B.

Hello I've been googling for a couple of days now, but still can't figure out what to put in zapata.conf to get it to report CID. Unless I'm mistaken, France uses ETSI FSK for CID method and bell 202 as CID FSK Standard: http://img219.imageshack.us/img219/7207/linksys3102cid1jj7.jpg http://img219.imageshack.us/img219/4625/linksys3102cid2ld5.jpg Does Zaptel support those on Digium

On Wed, Feb 17, 2016 at 8:14 AM, Mitul Limbani <mitul at enterux.in> wrote: > Sangoma 50 port FXS Thanks. Will I now stack 20 boxes in order to achieve the 1000 FXS lines? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20160217/0d4c2800/attachment.html>

Hi all, I need some advice - I have been working on originating multiple calls using AMI and then joining them. What I want to do is: - dial call 1 (where the caller is in a "channel" format, like SIp/1234 or Local/1234 at ext) and "park" it somehow - dial call 2 (where again the caller is in channel format) and join it to the previous call. As a requirement, I cannot use the

Thanks Mitul, The server spec is okay but I need information on the fxs hardware to use. Regards On Wed, Feb 17, 2016 at 8:07 AM, Mitul Limbani <mitul at enterux.in> wrote: > Quad core Xeon with 4GB ram > On Feb 17, 2016 12:32 PM, "Goke Aruna" <goksie at gmail.com> wrote: > >> Hello all, >> Can someone recommend what hardware to use for a 1000 analogue

Hello Everyone, I was wondering how SIGTRAN/SS7oIP can fit into our currently 100% SIP model. We are looking to interconnect with the PSTN world, and our supplier has given us a few options. We can either do this over traditional PRIs, A-Links or the SS7IP new. I am really interested in SIGTRAN, and was wondering how some of you have integrated it into your architecture. Can Asterisk handle

Hello, I have a fully functional Asterisk Server, I want to configure this server to be able to process call from Skype, can someone point me to a howto? or if there are suggestions on best way to approach this problem. Thanks, -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi there, I am using Asterisk 11.9 (with Sangoma-E1-Card/DAHDI) and it works fine :-) But I am wondering if there is a solution/application which will enable me to implement voice recognition while playing a voice file (barge in). So that the caller hears a voice file and can interrupt it with his voice. Currently (on our platform) the caller has to wait for the end of the voicefie. Then we play

Am looking for a service provider who can provide enterprise SIP trunk with 100 channels concurrent sessions. I see some like Inphonex, Broadvoice... and etc.... Is there any suggestions for the service providers. Regards -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, If anyone is interested, I have ported app_rtsp.c to Asterisk 11.x. I have tested it with GStreamer RTSP server and a C920 webcam streaming H264 SVC video from one machine to another machine running Linphone. Contact me at this e-mail address robkrakora at messagenetsystems.com for source code. Best Regards, -- Rob Krakora MessageNet Systems 101 East Carmel Dr. Suite 105 Carmel, IN 46032

Everybody, I've been using the old Asterisk CDR Areski GUI CDR-Stats for at least a dozen years, it was easy to configure and didn't requite installing 'connectors' on anything or adding tables on the DB server. It's based off of PHP5 and the only reason I still keep around a Debian 7 system, since it won't work with the newer PHP7. A friend of mine is learning PHP7

If SIP channel driver needs to connect to a remote GW over a dedicated SIP trunk BUT the remote GW has a 'standby' in case of failure, how can the sip configuration file be configured for the remote GW when there are actually two IP addresses. If the main remote GW fails control automatically switches to the standby GW, so how could the SIP configuration file hande this switch and support

Hi Helviom I am interested to evaluate your product. What asterisk version you build this product around? -- regards, abdul basit | p: +92 32 1416 4196 | o: +92 30 0841 1445 On Tue, Jun 23, 2015 at 7:34 PM, Tech Support <asterisk at voipbusiness.us> wrote: > Please keep the ?me to? emails off the list. > > Regards; > > JV > > > > *From:*

Is a normal functionality? when I do #dahdi_cfg -vvvvvv In my Asterisk console shows this.... [Jul 24 13:39:08] NOTICE[30263]: chan_dahdi.c:9461 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 If I do this a lot of times...then [Jul 24 13:39:20] NOTICE[30263]: chan_dahdi.c:9461 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 [Jul 24

On 04/07/2015 10:48 AM, Johan Wilfer wrote: > Den 2015-04-07 15:41, Ikka Tirtawidjaja skrev: >> Dear all, >> >> Is anyone has experience making Asterisk server with virtual server >> OPEN-VZ (in proxmox 3.4 box) ? >> >> My boss want to build a production server with it, and it will have +/- >> 300 sip user (concurrent call maybe < 150 call) >>

I'm looking into the dialplan specifics: tleilax:~ # tleilax:~ # cat /etc/asterisk/extensions.conf [general] static=yes writeprotect=no [globals] CONSOLE=Console/dsp ; Console interface for demo TRUNK=DAHDI/r1 ; Trunk interface TRUNKX=DAHDI/r2 ; 2nd trunk interface TRUNKIAX=IAX2/ASTtest1:test at 10.10.10.16:4569 ; IAX trunk

Hello Everyone, We are looking for a simple open source auto dialer with "polling" capabilities. What we would like is a program that we can upload leads to, and have asterisk: i) Dial numbers ii) Play pre-recorded iii) If user presses one, forward the call to an agent There are so many solutions out there it's hard to make a decision on what works, what has just a limited free

We have been using Red-fone foneBridge2 Quad T1/E1 for last few years. As these devices are not available anymore, we are looking for alternatives. Are there any similar devices available ? -- Regards, Tirveni Yadav www.udyansh.org What is this Universe ? From what it arises ? Into what does it go? In freedom it arises, In freedom it rests and into freedom it melts away. Upanishads.