similar to: AGI problema

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.'

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

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

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

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

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----

Hi everybody, I have a SER Server (Sip Proxy / REGISTRAR) and a Asterisk Server (PSTN and other services). I've got some clients that make calls to each other through my SER Server, that's to say, non external or international calls. I would like my clients to make external and international calls through my server but for that they must authenticate at another server to have a valid

Hi gentleman I've configured SER to forward every call starting with sip uri request "1" to Asterisk. I need to configure Asterisk as a Sip UAC in order to make it call to my other SIP Provider outside my network, sending username and password for authentication. I've read at www.voip-info.org some articles but found none that could suit to my needs, but yet I've found an

Hi everyone, I'm having some errors using Asterisk for incoming/outgoing calls. I have SER working with mysql without problems, all my internal users autenticate at SER and then if any number begins with a "1", Ser forwards the call to asterisk. Asterisk takes the forward and act as a Sip Client to make the call. I need to do that because My VoIP numbers are at Go2call, and I need

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

Hi, does anyone have the setup for go2call ? I have digium boards and quicknet linejacks and phonejacks. The cards work fine in asterisk without the g729 or g723.1 for the phonejack. I will like to do SIP origination using the codec in the phonejack and linejack g729 or g723 and send the calls to go2call. Anyone has the setup for this ? Or similar setup to a SIP provider using g729 or g723

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

hi there, I have just downloaded and installed asterisk a couple of days ago, it compiled correctly and starts up and runs, on a Redhat 9 system freshly installed for testing. I don't have any extra hardware installed so far, was attempting to just try out connectivity. I am having some probs with the configuration, maybe someone out there can give me some tips : firstly on modifying the

I'm configuring my SER to forward calls based in extension. Cause I would like my ASTERISK to do international calls. How could I make ASterisk do international calls ?? I must pass the host (Go2Call), username and password to get the call up, but I don't know how. I'm trying to find a extension command that like Dial, does the call but passing username, password and host for

https://bugs.freedesktop.org/show_bug.cgi?id=87934 Bug ID: 87934 Summary: nouveau errors on archlinux while running War Thunder native (not via steam) Product: Mesa Version: unspecified Hardware: x86-64 (AMD64) OS: Linux (All) Status: NEW Severity: normal Priority: medium

Hi, I'd like to establish multiple direct connections between 2 hosts. As far as I can see, tinc doesn't allow this (when a new connection come, the old one is closed). Is seach features maybe integrated to tinc ? What is the status of 2.0 svn branch ? Thx. Manuel

When I read variables in AGI scripts, I see only the follwing 13 variables agi_request agi_channel agi_language agi_type agi_uniqueid agi_callerid agi_dnid agi_rdnis agi_context agi_extension agi_priority agi_enhanced agi_accountcode beside these, I found following variables documented on several sites. agi_calleridname agi_callingpres agi_callingani2 agi_callington agi_callingtns Where can I

Hi, I have been written an small script for test the use these commands. I had done massive test with commands, but I didn?t get success it. Any of the cases, I don?t listen nothing on channel that call 2100 extension. I dial 2100 extension through an cisco phone 7912 with SIP, also I dialed through ATA SIP (Linksys PAP-2). I?m using Asterisk 1.2.7.1 and ztdummy driver, linux kernel 2.6.11.4. I

Hello, I am a beginner in using R so my question could seem very simple. I would like to download the package multiv to do multivariate data analysis. The package I download seems to be a file meant for UNIX and I am using a Window OS. How could I download and install correctly this file? Thanks a lot _________________________________________________________________ MSN Messenger :

Dear friends of R, I want to produce a pdf file with the contents of a matrix. I employ the latex command in combination with dvi, both contained in the Hmisc package. It seems to me that the function does not correctly set the directory. > tbl.loc <- matrix(1:4, nc=2) > latex.obj <- latex(tbl.loc) > dvi(latex.obj) warning: extra args ignored after 'cd' H:\PROJECTS\data