similar to: Voicemail / Dial command issue

Quick question from a newbie, I have asterisk configured to dial IAX extensions (which works). When dialing from one IAX extension (using Firefly) to another IAX extension (also using Firefly), the Firefly client rings on the receiving end and gives the option of accepting or denying the call. However, when I dial in to Asterix using a VoicePulse number and dial the same extension Firefly

Trixbox 2.2 Beta with freePBX 2.2.0rc1 I have a setup that looks something like this in ASCII art: Teliax IAX Trunk ------+ | V Embarq PRI ----> Tandem switch ----> Ottawa Office Server------+ +--------------> Lima Office Server -----+|

Hello! I have a number of IAX clients behind a NAT (on the same LAN) and asterisk server on the Internet. And that clients doesn't speak directly to each other, traffic goes through the asterisk server. What should I configure to make IAX clients on the same LAN to speak directly, please? notraster=no is set in iax.conf The asterisk server is on real IP behind a NAT (at DMZ with full 1-to-1

Dear all i'm trying to originate an outgoing call with the command originate, from Asterisk's CLI i'm typing: CLI> originate IAX2/my-iax-provider/number2call application wait 10 [Apr 18 19:31:12] DEBUG[32331]: chan_iax2.c:4000 create_addr: prepending 40 to prefs -- Call accepted by 62.149.202.150 (format ilbc) -- Format for call is ilbc -- Hungup

Guys. Im using IAX and FWD and I think everything is setup fine.. someobdy just tried calling me but my phone jus ran once and sent them straight to the voicemail.. the logs show this: -- Accepting AUTHENTICATED call from 65.39.205.121: > requested format = ulaw, > requested prefs = (), > actual format = ulaw, > host prefs = (alaw|ulaw|ilbc|gsm),

Hi, This problem may be related to a configuration problem but I believe it is a bug in the FOP since restarting the FOP server clears the problem. Here is the scenario: Using AgentCallBackLogin and have four agents logged in a call is made to one of the agents directly from an internal phone. Okay so far. Call is hung up and the same extension is used to call another agent okay again, no

Dear All, I tried to use 'L' option on my dial command. I set the x to 65000(65 seconds), y to 60000(60 seconds), z to 30000(30 seconds). The max calltime should be 65 seconds, and it will play "beep.gsm" at 60 seconds left. And repeat the beep at 30 seconds left. But the call will be hangup by system at 60 seconds left. In another word, when it plays warning file, the call

Don't know enough to properly term the problem I'm seeing... sorry if subject appears vague. And I have other questions too, but "Newbie Help Wanted" isn't exactly more specific... ;-) My setup, intended for testing and all, "*" version 1.6.0.6, dahdi with an X100p clone. Regular phone line provides PSTN access with one port (and my DSL). Calls come in and are

I have something going on that I don't fully understand after a weekend of looking for answers. I have an iax account with Tesco that works flawlessly with the Zoiper client - but is giving me trouble with inbound calls in Asterisk 1.6. After some playing I have ended up with an iax.conf file that looks like this: [general] calltokenoptional = 77.75.0.0/255.255.248.0 maxcallnumbers = 16382

Hi all, I'm a newbie and I have a configuration problem with Asterisk. Seems that I'm not able to call an outbound number. I'm quite sure that it is a configuration problem, but I'm not able to find out where is the mistake, even reading several docs to www.voip-info.org. I do not have a good knowledge of Asterisk, I'm not very familiar with its configuration and I've a

Dear list; I'm trying for forward some calls to an others asterisk using IAX2 protocol. But My asterisk can forward some calls and for others it hangs up automaticaly. Before my asterisk was working perfectly, i do not know what is happening!! When i try directly zoiper with my provider's asterisk it works perfectly. Here is the output from the cli when i made a call that asterisk hangs

Hi Here is part of my extensions.conf exten => 8661231234,1,agi,dtmf.php When I dial this number, this is what I see in my asterisk console: -- Accepting AUTHENTICATED call from 198.22.67.70: > requested format = gsm, > requested prefs = (), > actual format = gsm, > host prefs = (gsm|ilbc|speex), > priority = mine -- Executing

I am having problems getting any form of call transfer working. I have reconfigured blind transfers to be #1 and assisted transfers to be *2 but these are not working. Looking at the wiki (http://www.voip-info.org/wiki-Asterisk+cmd+Transfer) it it does not mention IAX so I assume I have to use the native IAX transfer supported by Diax? I have tried using Diax but am getting a problem that after

When I try to dial out I get an error: Operating with different codecs [0x2 (gsm)] [0x4 (ulaw)] , can't native bridge.. Here are the details: -- Accepting AUTHENTICATED call from 66.18.210.217: > requested format = gsm, > requested prefs = (), > actual format = gsm, > host prefs = (gsm|ilbc|ulaw|alaw|speex), > priority = mine --

I have a Sangoma Card with two PRIs. They are both configured in Zaptel and Zapata; In Zapata I have them separated in Group 1 and 2 but if I make a call and specify Zap/g2 it doesn't go when calling Channels : HERE IS what I get: Accepting AUTHENTICATED call from x.x.x.x > requested format = speex, > requested prefs = (), > actual format = gsm,

Hi All. I have many days reading and research about asterisk and vicidial. I thing this issue is about asterisk and doesnt about vicidial. Isn't it? I have a problem with theses application (I already ask for help in vicidial forums), but I can not fix it. I have debian 5 with asterisk 1.2.24 and vicidial 2.0.4. This server has a IAX tunnel with another asterisk server B which connect to

I tried to update and build 1.4 (SVN-branch-1.4-r45775) tonight. I took the additional step of nuking my modules directory first... When I used the command asterisk -v to start asterisk, it seemed to go along and get to the point where asterisk is running(ie Asterisk Ready). At that point it was eating all available CPU. I went ahead and tried to register a softphone to it via IAX2, which

Hi all, I am trying to understand how I can get a simple IVR scenario to work properly (having already removed most of my hair...). The basic requirement is as follows: * Caller arrives at our main number * Caller is greeted and then told they can enter an extension number, if known, or wait and their call will be connected to an available rep. * The IVR then dials a group of extensions (if

I need to debug a call quality issue with remote users on the other end of a satellite link. The symptoms are: we here on the Internet side can hear them just fine. On their end, things work sorta OK most times, but they often suffer from severe dropouts and digital warbling, both of which I attribute to them missing packets. Often times they can't make out a word we are saying while we can

Have asterisk@home 1.2.1 The server is on an internal network eg 10.10.10.10 It is NAT'd 1:1 via Checkpoint firewall to external public IP eg 50.50.50.50 The remote IAX2 phone (ATCOM320) is configured to call 50.50.50.50 on extension 1055. Outbound calls to 1055 work perfectly. Inbound calls from 1055 get picked up as if it were an external call (see below) and goes straight to the ring