similar to: asterisk wait for traling digits

I am trying to set up a Grandstream GXV-3000 Video phone to Asterisk ver 1.2.21.1. The problem I'm having is that it can call other SIP phones, but not vice versa. Can someone tell me where is the problem? TIA! Here's part of my configurations: ---------- sip.conf ---------- ; 113 is the Grandstream phone [113] type=friend username=113 secret=secret context=default dtmfmode = rfc2833

Hello... I have a big problem with asterisk. Every time i make a call asterisk does not bridge the zap channels. The zap channel from which i'm calling remains in state:ring and applicaton:dial and the zap channel with the external line configured remains in state:dialling an Application:AppDial. Zap/20-1 agentie s 1 Dialing AppDial (Outgoing Line) 09399 (None) Zap/9-1 int_omg 09399 5 Ring

Hi List; Can someone advise me why in the below context, it does not run the Background step? Once I dial 1000, then it hangup and give congestion signal? If I comment the ResponseTimeOut, then it run the Background but it does not wait till caller enter the digits, once the sound file finish, then it hangup (congestion signal), also in all the situation, it does not go for the t extension, why?

Hi, I have problems while trying to dial from simple analog phone that attached to my TDM400P card. No matter which number i press i immediately get a congestion tone. when calling from outside (e.g cellphone )to the line on port 4 and pressing extension #123 everything works fine and i manage to make a connection. I've plugged on port(Zap) 4 the analog line and on port 1 the phone.

Hi Guys, I am not so familiar with asterisk and hope to get help here. I am having now some stupid errors. My goal for the first, is to create a simple pbx with different context. As long as I use only the contex 'default' everything seems to work perfect. Now I tried to add another context i.e 'internal' and the asterisk is complaining for not finding the required extension in

Dear ALL I have audiocode MP -124 with configure in asterisk Endpoint configuration means every analog phone register in asterisk now thing is that i have one more SIP trunk with mediant 2000 [auodiocode-mp-124]-----[ * ]------[mediant 2000]-----E1 When i call from audiocode MP -124 phone i got this error -- Executing Dial("SIP/20-0889c4d8", "SIP/mediant/1")

Hi All, i want to configure voice mail on Asterisk 1.4 for multiple users. let me explain you the scenario. i have 10 users with the name of 1000,2000,3000,4000,5000,6000,.......and these user can call to each other. Now i want to configure separate voice mail box for separate user. my extensions.conf ..... settings below.. [voicemail] exten => _X.,1,Dial(SIP/${EXTEN}) exten =>

Hi All, i want to configure voice mail on Asterisk 1.4 for multiple users. let me explain you the scenario. i have 10 users with the name of 1000,2000,3000,4000,5000,6000,.......and these user can call to each other. Now i want to configure separate voice mail box for separate user. my extensions.conf ..... settings below.. [voicemail] exten => _X.,1,Dial(SIP/${EXTEN}) exten =>

I tried to add a couple of SIP phones (polycom 601s) to my existing asterisk installation. I can successfully make a call from the SIP phone to any other phone (inside or outside), but I can not make any calls to a SIP phone. Attached are the pertinent parts of sip.conf and extensions.conf. The log starts off normal with: Mar 26 09:51:15 DEBUG[4885] chan_zap.c: DTMF digit: 2 on Zap/55-1 Mar

Dear all I have Asterisk which is connected with avaya through E1 back 2 back now i have on asterisk side G.711 codec and Avaya also useing G.711 codec everything fine. I need G.729 on my asterisk side. can i have lots of SIP phone on my lan and issue is i have 2 to 3 building so problem is LAN is congested thats why i need G.729 Now testing perpose i have download

Hi everybody, Well, I've finally got asterisk to to talk nicely with my Intertel pbx. Currently there is a outside T1 line (e&m wink start, esf, b8zs) connected to asterisk, and then asterisk connected similarly to my Intertel pbx. For right now all asterisk is doing is passing calls between the two. When I call out from the pbx, I can connect perfectly to the outside world. When I

Hi list, We're trying to set up a phone number that customers can call to get a hold of anyone of a group of sysadmins (and not their voice mails!). We found the findme example ([1]) that makes the callees press 1 to accept the call. It almost works, but it doesn't work correctly when one of the callees, the sysadmins, hangs up after accepting the call. We're using this

Hi all, i am stuck with the configuration of asterisk - modules are loaded ( zaptel and wct4xxp ) - i have zaptel.conf configure, output of ztcfg -vv --- snip -- rapid:~# ztcfg -vv Zaptel Configuration ====================== SPAN 1: ESF/B8ZS Build-out: 0 db (CSU)/0-133 feet (DSX-1) SPAN 2: ESF/B8ZS Build-out: 0 db (CSU)/0-133 feet (DSX-1) SPAN 3: ESF/B8ZS Build-out: 0 db (CSU)/0-133 feet

Hello group, I tried a lot to enlarge the frequency (i.e. more announces, low wait between). according to config, every 30 seconds the announcement should take place. In fact, the first periodic announce is done after 2 minutes? What is my fault? Thank you Regards Paul # zypper if asterisk Loading repository data... Reading installed packages... Information for package asterisk:

I have Asterisk running with a combination of SIP and H323 clients. I am using the OH323 module instead of the H323 one. When the SIP clients ring each other, they can hear a ringing noise in the ear peice to let them know that the other parties phone is ringing. However, when the H323 client rings a SIP client, there is no ringing sound at all, although as soon as the called party picks up the

Hi, I run into a problem and I'm not shure what do I misconfigure. I've a B410P ISDN card with bri_cpe signalling and two Openvox (A1200, A800) cards with fxo_ks signalling, all with dahdi drivers. I can receive fax from a public number, but I can't send fax. The CLI says it picks up the line but no dialing. I tried the extension with an analog phone, it works fine, I can dial

Hello everyone, I've just upgraded from 1.6.1.9 to 1.6.2.13. I have noticed that (after the upgrade) abandoned calls within the Queue produce no cdr at all. I am using unanswered = no (the default) in cdr.conf. The call shows, as expected, in the queue_log as ABANDON The dialplan is: Ringing(); Answer(); // need to answer or no music! goto

Hello all, I have been posting some questions about this problems that I cannot yet solve, but I think I have a better diagostic, so maybe someone can give me a clue why it is happenning. I have Asterisk + AMP configured as a PBX with a Customer Center Queue with 4 agents that login/logout dinamically. If there are no agents, queue timesout and gets derived to another queue that somebody

Hello,For the time being I am using the following line to play the original saved message by Asteriskexten => 7001,n,Playback(vm-nobodyavail)Now I am trying to use the other features for Asterisk's voicemail. I have recorded a message, and I can see it saved on the system, but still Asterisk keeps playing the original message... Is there something I can add to let the subscriber plays his

I want to be able to pass any number (variable length) to a context and then forward that to another asterisk server for processing by that servers dial plan.? I have the two talking IAX2 so that part is done. I can also dial a number from the sending to the server asterisk. The problem is I don't want to have to create (duplicate) dial plans at originating Asterisk to equal those at the