similar to: Wildcards and variable number of digits

I have looked all over the site(s) for help. But heres the problem. Im missing something. In coming works fine from FreeWorld via IAX. But when Dialing out i get: May 13 13:42:01 WARNING[1150495040]: chan_iax2.c:5256 socket_read: I don't know how to authenticate iaxtel to 65.39.205.121 my IAX.conf if as follows [general] port=5036 register => ######:xxxxxxxxxxxxx@iax2.fwdnet.net

Greetings all, My next step is to purchase a nice VoIP phone for my desk. I have a grandstream, and the sound is great, but I'm looking for more of an office style phone, preferably that can handle multiple lines, has a more flexible display (i.e. name as well as number). SIP would be preferable. Any suggestions? Thanks, Eric

============= SJphone Log ============ Outgoing SIP session Respondent: (sip:8612@192.168.2.2) Remote client: Started: May 26 16:33 Accepted: no Ended: May 26 16:34 End reason: Call rejected: 503 Service Unavailable =============== Asterisk Debug ================ Executing Dial("SIP/2201-a83e", "IAX2/<FWDNUMBER>:@iax2.fwdnet.net/612|60|r") in new stack --

Does anyone know of a company that provides German DIDs (preferably Berlin) and termination of calls to Germany at reasonable rates? Thanks, Eric jacksch@tenebris.ca

Hi, I am setting up an asterisk server for testing purposes and cannot get voicemail to work at all. My host OS is Linux From Scratch 6.3 and the asterisk software versions I built are zaptel-1.4.5.1 and asterisk-1.4.12. I am using the Ekiga softphone on my Ubuntu desktop machine. My asterisk server and client phone are on different computers but are on the same LAN, i.e. no NAT. I have an

Hi, How could I possibly inform incoming callers that the number they'd dialed is monitored and recorded. I wanted that when a call-in or call-out is made, a playback will be played to inform caller & callee that thier line is monitored prior to start conversation. Thanks. Angel __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best

Hi to all the * people out there, Please kind to me as I am both new to Asterisk and to Linux - But I am learning fast. My config is quite simple, I'm just following examples and the Wiki: I have two PC's running X-Lite phones, these work without problems between each other, and I have a GS BudgeTone-100 registered to Free World Dial UP (working no problem). I have tried to

Greetings all, I've tried out chanspy, but what I'm really looking for is the ability to interrupt a call (i.e. barge in for emergency purposes). Has anyone found a way to do that with Asterisk? Regards, Eric -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060403/dfe6e99e/attachment.htm

Greetings all, I've been working on having Asterisk put a call through to two different numbers, and give the call to the first one that acknowledges by pressing the 1 key. I found an example on the wiki, but I can't get it working. When I answer the call I hear the message telling me to press 1 to connect, and as soon as the message is done, the call is connected. In other words, it

I have configured iax.conf and extensions.conf as instructed on FWD website (http://www.freeworlddialup.com/help/?p=knowledgebase&c=18&a=76) and I can successfully receive calls and make test calls to 612, 613, etc. The problem is that I can not make a call to another FWD user. Here is what asterisk says: -- Executing [393xxxxxx@default:1] Set("Zap/1-1",

When I use DISA I get congestion when I try to reach 1-800-number: Here is the context: [disa] exten => 087,1,Answer exten => 087,2,DigitTimeout,8 exten => 087,3,ResponseTimeout,20 exten => 087,4,Authenticate(985) exten => 087,5,DISA(951|disa-access) [disa-access] include => tollfree include => outgoing-voipjet [tollfree] ; ; terminate toll-free no.'s via fwdnet ; US

I'm simply trying to forward calls to users who have the call forwarding feature enabled (FWD Status and FWD Ph Number kept in the astDB). The problem is that I want users to be able to forward calls to numbers that they would normally be allowed to dial within their own context. (I don't want a local call only person forwarding to a long dist number, for example.) I'm able to

I am having trouble with this. I can dial 1800 numbers fine as well as FWD service numbers but not Vonage. I can be called from ipkall and fwd and can call aixtel numbers. I use aix2 with Fwd. My extensions.conf for Vonage: ; vonage numbers ; ; +2431 exten => _2431XXXXXXXXXX,1,SetCallerID,${FWDCIDNAME} exten =>

Perhaps this will help... I have a phone connected to a QuickNet PhoneJack card. When I pick it up, I get a dial tone. When I dial a certain number of digits, the call is processed by Asterisk. The question: How does Asterisk determine how many numbers to let me dial? I'm banging my head against the desk here... _9XXXXXXX lets me make an outbound call, but _9X. only lets me dial 9 plus

Guys. Can you dial 800 and 888 toll free numbers using FWD? how do you dial them cause I tried using 1800xxxxx and 1888xxxxx and I simply get a "nobody can asnwer the call" signal on asterisk. Can you dial 800 toll free from FWD?

An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040802/d9eed8fc/attachment.htm -------------- next part -------------- ? Hi Can anyone give suggestion why we need STUN while using asterisk behind the NAT. Regards Shan.

The problem I am running into is that when the call to my cellphone is made, it appears as though the call "completes" so it never rolls to asterisk voicemail. Here is my current config: exten => 102,1,Dial(${sipura},10,) exten => 102,n,playback(pls-wait-connect-call) exten => 102,n,Dial(IAX2/asterisk1/9139275900,10,r) exten => 102,n,VoiceMail(u102@default) exten =>

Hello, I am trying to setup my Asterisk box to accept calls from the FWD network. I've followed all the config advice / samples I've found on the web. Making calls to devices on the FWD network from my Asterisk box works flawlessly, but whenever I try to call my Asterisk box from a FWD client I get a busy signal, and a "Call Disconnected" 486 error. What's odd is that I

Well, I thought I had my problem solved, but it is acting up again. Hopefully this time I can provide enough information. What I have is an * box setup with one X100P and TDM400 with one FXO and one FXS. For my regular setup with interfacing with my PSTN and my entire house with analog phones, the box is working great. I am trying to interface a Mediatrix 1202 device to my * box via the

Hi, I've been trying to dial out to a cellphone, but all my calls get redirected to 066 (the emergency number at my city, like 911) does anyone know how to fix this, any ideas,? does anyone from mexico has done this? Any comment will be highly appreciated, Regards, Claudio -------------- next part -------------- An HTML attachment was scrubbed... URL: