similar to: TDM card and voicemail volume

I have a user who goes into Comedian Mail for the first time and goes thru the initial setup, changes password, records name, etc. Problem is that every time he calls in, it thinks that it's his first time and keeps reprompting him. His password change is reflected in voicemail.conf. Others do not have this problem. Where does Asterisk maintain the "first time" flag? Any ideas

I just set up a Polycom 500 on *. Every few calls I make, the call connects and the receiving party can hear me (thru Broadvoice), but I still get ringing on my end, as if they never picked up. * logs look just fine. Does any one have any suggestions? Thanks. ________________________________ Adam S. Robins Executive Vice President & CIO PHARMACENTRA, LLP 5901B Peachtree Dunwoody

Can anyone out there please tell me what ports Speex uses? I want to set up QoS on switches but I can't seem to find this information anywhere. The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent in trust, for the sole purpose of delivery to the intended

Hello, Being totally fed up with the lack of quality and reliability from both VoicePulse and BroadVoice, We are switching to a direct IP connection to Global Crossing. We've installed a local point-to-point T1 into their CO, and they will give/take SIP g729a directly and act as the gateway for us. In setting up the inbound SIP service, they are asking the question, "In what format do

I'm trying to compile the voicemail module with mysql support and I get this error on the chan_zap module . Does anyone have any idea's on this one.. chan_zap.c: In function `handle_init_event': chan_zap.c:5668: error: `ZT_EVENT_POLARITY' undeclared (first use in this function) chan_zap.c:5668: error: (Each undeclared identifier is reported only once chan_zap.c:5668: error: for

Is anyone else having problems with them? Until today everything was working fine. But now dtmf is not working on incoming calls. Any ideas? I tried calling them and their voicemail is not accepting answers. Is there another source for DIDs in the 314 or 636 area codes? Especially a company that supports something besides ulaw. I am going to hate switching numbers again, my wife is

We are in the process of an Asterisk call center deployment using IAX2 G711 ulaw softphones. Outbound sound quality is terrible. This week we rebuilt the entire LAN with Cisco 2950-EI switches and have employed QoS on the switches and router. Still sounds terrible. What we are now finding is that the network card in the PC may be the key to the problem. A Dell Optiplex P4 2.4GHz 512MB

I have a real head scratcher . . . We have several employees who work from home. All have Polycom 501's that register to our office Asterisk 1.6.x server and communicate using SIP g729a. About two weeks ago, one of these remote users starting experiencing a problem with a previously working phone: a. She could receive inbound calls, b. She can place outbound calls to internal extensions c.

Anyone out there using small-midsized (2-4 TB) SAN solution among multiple Asterisk systems? I don't have the budget for an EMC-caliber solution, and can't seem to find much else out there. Thanks, Adam The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent

I am attempting to update my Asterisk installation from 1.0 to the latest stable version. When I use CVS checkout, I am receiving the following messages on chan_sip.c: RCS file: /usr/cvsroot/asterisk/channels/chan_sip.c,v retrieving revision 1.510.2.25 retrieving revision 1.510.2.27 Merging differences between 1.510.2.25 and 1.510.2.27 into chan_sip.c M asterisk/channels/chan_sip.c Then, when

Does anyone have any recommendations on implementing Answering Machine detection for call generation programs? What I would like is * to determine what picks up the other line (Answering Machine, Voicemail, or Human) to determine which action to take. For example: If * detects Answering Machine or Voicemail, hangup call & the AGI will log (ANSWERING MACHINE DETECTED) and at that point,

Does anyone know of a really lightweight web interface that allows users to log in and modify attributes of their extension only? Thanks Curt -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061030/892a67b2/attachment.htm

I am switching from IAX2 to SIP for my inter-Asterisk transport due to assorted quality issues following the 1.2.4 upgrade. On the server that SENDS the call, I have the following in SIP.CONF: [192.168.1.2_OB] type=peer fromuser=OB host=192.168.1.2 And in EXTENSIONS.CONF exten => 91NXXNXXXXXX,1,Dial(SIP/${EXTEN}@192.168.1.2_OB) On the RECEIVING Server in SIP.CONF: [OB] type=user

Hi all !!! I would like to install asterisk in Xen domU using TDM400 hardware. Somebody know a howto or tutorial about that ? Thanks in advance roberto -- Ing. Roberto Pereyra ContenidosOnline http://www.contenidosonline.com.ar

Using Digium's "Asterisk Developer's Kit (TDM)", I've been trying to make an outside call by copying sample.call to /var/spool/asterisk/outgoing. I want the VoiceMailMain to run when the call is answered. The call is made correctly but, as you probably know, the application starts as soon as the call is made. I see there are two solutions: Using callprogress=yes in

Hi, everyone: I am developing a system using Asterisk, TDM-400 analog cards, analog lines, and Polycom SIP phones for internal extensions. Initially there was bad echo but after a series of efforts, I've managed to reduce it to a negligible level (it only happens when both parties speak simultaneously, and even there, only for a few hundred milliseconds). From an echo standpoint, things are

I was using G729 with Asterisk 1.07. With the new ability to do packet loss correction with ILBC, I felt I'd give it a try. The new PLC does not work with G729. I don't use Speex because my softphone does not support it. This is a 1.5mb IP-VPN connection with prioritized QOS for port 4569 (IAX2). I've never really stressed the bandwidth. Typically, only 10-20 concurrent calls.

Hi, sorry for my insistence but this is a big problem for me..my steps for remove card are ok ? Thanks. ------ Salvatore. ----- Original Message ----- From: "Sasa" <sasa at shoponweb.it> To: <asterisk-users at lists.digium.com> Sent: Monday, November 26, 2007 4:25 PM Subject: [asterisk-users] Remove a TDM Card > Hi, I would like remove a Digium TDM2400P from

Hello, I have set up Comedian Mail on my Asterisk system. I am using Voicemail not Voicemail2 in my extensions.conf file. The system works great except for 1 thing...It is not possible to create custom unavailable or greeting messages for 3/4 voicemail boxes. For some odd reason 3/4 users are unable to modify the default voicemail prompt with their own custom greeting. The greeting gets

Has anyone made spandsp to work with a digium tdm fxo card? I finally got the rxfax and txfax modules to compile, the spandsp lib installed (and in the libpath), and now receive: -- Starting simple switch on 'Zap/1-1' -- Executing RxFAX("Zap/1-1", "/var/fax.tif") in new stack -- Hungup 'Zap/1-1' I've tried to adjust rxgain/txgain in