similar to: beginner's question!

hi, maybe someone out there already has some experience and can help me. I have just ordered an E100P card from Digium, I already have a basic asterisk setup up & running. My application is the following : I want to accept incoming calls from the PSTN to Asterisk, and without asking anything of the client just pass them immediately to a call gateway in USA, actually we are planning to use

i've been having a problem getting two SIP phones to bridge running through asterisk, actually one is a SIP softphone, SJ Phone, and the other is the Go2Call calling gateway. Someone suggested that I don't have the right codecs. How do I find out which codecs are installed, and how can I install further codecs? Any suggestions which would be the right one? I think hte problem is from the

I added the line "allow G723.1" in my sip.conf general config, and from a bridge connection which gives silence, I have progressed to the error message below, and the call gets rejected. help!! Dave ps. 217.168.168.49 : soft sipphone, i'm trying SJphone & Pingel Instant Expressa 723@216.52.153.207 : Go2Call SIP gateway -- Executing

hi guys, I have a little problem maybe you can help ... I have an asterisk setup, with an E100P, and an ISDN-PRI 30 channel line from the telco going into it .. the E1 line is OK, because plugged into a Lucent Portmaster 4 it works OK .. plugged into the asterisk box I just get an engaged tone, and asterisk posts this message on screen : WARNING[1167272000]: File chan_zap.c, Line 5275

Hi all, i use the DISA app for giving the user a trunk after a authentication through PGSQL as follows .... auth via PGSQL exten => s,1,DISA,no-password|test I think the user is now in context "test" and he could dial any number if the extension-conf in "test" is for example exten s,1,Dial,OH323/<myip> But if the user dial one digit the call build up

I am trying to get asterisk to dial this address : sip:723@216.52.153.207 Using a softphone on my PC (217.168.168.49) it dials immediately and I get a voice prompt .. I have configured an extension, 1303 on asterisk, modifying the demo configuration : exten => 1303,1,Dial(SIP/723@216.52.153.207) When from my softphone I dial sip:1303@217.168.168.51 on the console I get : -- Executing

hi .. I have an asterisk system with three TDM100P (single port FXO) cards and 10 Grandstream 100 phones connected to it .. 1st question: when i phone out or receive a call from one of the SIP phones onto the PSTN, there is a LOT of local echo in the handset .. the PSTN end of the call does not here this echo, but it's VERY annoying on the SIP end of things .. the echo seems to be about 0.3

hi .. i've just installed and licensed an instance of the G729 codec. I am trying to connect through asterisk to Go2Call server .. According to their info it involves dialling extension 729 on voip01.go2call.com, to get the IVR. my extensions.conf shows : exten => s,2,Dial(OH323/h323:729@216.52.153.206) which I think is correct, I have G729 enabled in the OH323.conf file and it seems to

just three "stupid" questions I need to ask .. 1. what's the sequence to press on a SIP phone to transfer a call to another extension. 2. what's the same thing if you want to hold an incoming call, speak to the other extension, then pass the call? 3. what's the extensions.conf syntax to dial two SIP extensions at once? many thanks Dave

I'm having trouble connecting a asterisk server to a SIP Express router. Inbound calls to my asterisk server works just fine, but when i try to make outbound calls I get the following error message: Nov 27 22:40:48 NOTICE[4687]: chan_sip2.c:7967 handle_response: Failed to WWW-authenticate on INVITE to '"username" <sip:username@mysipprovider>;tag=as5399a078' I'm

Any got experience with these? I couldn't fint anything in any postings... it seems they have a h.323 on voip01.go2call.com and a sip on sip01.go2call.com I have tried to register with some of the same as I use for nikotel, but Asterisk does not want to register. I've tried to use both the user name (ingvald) and the PIN code 440.... as authentication. ---from sip.conf----

Can asterisk act as a SIP registrar or location server? I would like to be able for a user agent(client) to register with whatever client they are using as "username@domain-name.com". Rather than the entry/username/password that is setup in the sip.conf file. That way a user could log into any SIP enable client and their calls would follow them around. I have read the sip.conf man pages

<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN"> <html> <head> <meta content="text/html;charset=ISO-8859-1" http-equiv="Content-Type"> <title></title> </head> <body bgcolor="#ffffff" text="#000000"> <font face="Verdana">Hi all. I've written a AGI in C language.

I have just compiled & installed the latest oh323, on a fresh asterisk installation however using a previously working oh323.conf file. When I try to dial an outbound oh323 call I get the following error : -- Going to extension s|1 because of immediate=yes -- Executing Wait("Zap/1-1", "1") in new stack -- Accepting call from '21382890' to 's'

i've just lost $2000 dollars or so on my first commercial asterisk installation .. i'm running a PIV class server, three Digium Wildcard FXO cards, and 10 Grandstream Budgettone SIP phones. The system was to be a PBX for a small company. After over 2 months of pissing about, the client has had his fill of asterisk problems, and asked me to take my equipment out of the building. Obviously,

hi .. i require to interface asterisk to a 60 line analog PBX in a hotel. I was thinking of giving Asterisk a couple of PBX lines interfaced through cards, and then place outgoing calls through SIP/H323 and a DSL connection. analog extension lines <--> analog pbx <-->asterisk <--> SIP --> termination I do not need incoming calls to the lines. My question is this : if I

Hi Everybody, I'm trying to make my asterisk dial a international call from a SER request of it. My ambient is like this. [Clients]--[SER]--[Asterisk]--[Go2Call] Client: My SIP clients. SER: My REGISTRAR/Proxy Server Asterisk: All other services(Voicemail,musiconhold etc) and also acting as an UAC dialing International Calls, because SER doesn't do that sending username, password and

If you update your source from the CVS, you'll get a new SIP channel that supports a new syntax for SIP calls in extensions.conf If you define a SIP peer in sip conf, like [mysipprovider] ... You can now use dial(SIP/mysipprovider/extension) Where the part "mysipprovider" is related to the sip.conf section. Also, you can dial any SIP URL by

I'm trying to understand IAX and whether or not it would solve my difficulties: 'The primary goals for IAX were to minimize bandwidth used in media transmissions, with particular attention drawn to control and individual voice calls, and to provide native support for NAT (Network Address Translation) transparency. Another goal is to be easy to use behind firewalls.'

Hello all: For fun, I am learning about Asterisk, and trying to get Asterisk working at my house. I installed Asterisk@Home. It seems to be functioning fine. I installed a couple of softphones, and have them registered with Asterisk. I actually work for a CLEC, and I have registered my Asterisk box with SER (which I don't begin to understand yet) at the office. In order to try to