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Hi people. The latest version of my Ethereal plugin for IAX2 is now available here: - http://almaw.com/ethereal-iax2-plugin-0.3.zip A screenshot showing what you're missing is here: - http://almaw.com/ethereal.png The new version adds the following features/bugfixes: - Decomposes the CODEC fields for supported CODECs, complete with nice English descriptions. This gives you a list...

As mentioned on the devel list earlier today, I'm interested in writing an IAX2 plugin for Ethereal to make debugging IAX protocol implementation and simultaneous calls on normal networks easier. Anyway, I started work on it this evening, so it's not complete yet, but it's starting to look quite sensible: - http://raq626.uk2net.com/~al/ethereal.png A couple of people have

Lots of people seem to want this, so I've stuck it up here: - http://almaw.com/ethereal-iax2-plugin-0.1.zip Note that it currently only does IAX-2. I might expand it to cope with IAX-1 at a later date, but no promises. It's fairly basic - unzip the file and follow the README instructions. Regards, Alastair

Hi all, I am trying to write a program that sends SIP requests to asterisk. My aim is to make asterisk record as many voicemails it can at a time. The design of the program is like this: There are two processes: One main process and a child process (No flames pls. I have very little idea about pthreads and dl modules) The main program asks the user to input the number of test instances. When

hi see below's conversation. it seems the e100p card doesn't work with BT. Any idea how this'll work against Telenor (norway)? roy <RoyK> does anyone know if I can trust the E100P to do full PRI stuff in .no? <cypromis> dunno about no <cypromis> I cannot use it in UK <cypromis> cause the framer has problems with system-x switches at bt

I can't seem to hide and/or set my caller ID from *. I'm using a quite recent (three weeks or so) CVS with an E400P card. I have pridialplan=unknown in zapata.conf and I'm based in the UK. The relevant bit of pri debug looks like this (reformatted to fit 80 char width): > Calling Number (len= 4) [ Ext: 0 > TON: Unknown Number Type (0) >

Beginner: How can a script be called after a calling user hangup? What's wrong with this: [incoming] exten => s,1,Playback,welcome exten => s,2,Record,msgfile:gsm exten => h,1,Goto(callscript,1,1) [callscript] exten => 1,1,Wait,5 exten => 1,2,System("SomeScript") Thank you -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, It would be great if the IAX protocol will be able to tranfer fax data (even converted in another format) between Asterisk boxes, using low bandwidth codecs like GSM. I know that this is possible only with the G.711 now (passing faxes using the audio stream), but.... maybe in the future...some native support will permit this. Just a thought, Dan

Hello all. I'm new to asterisk and have been using and testing it for about a week now. My initial hope has been to use it as a sip<->h323 gateway to tie SIP & H323 based ip phones together with my Cisco AS5300 and Lucent MaxTNT/MVAM networks. I am currently running Asterisk 0.5.0 under Redhat 9 on a single PIII 800 with 256megs RAM. I have tried a couple CVS version from the past

Hi, I have * set up as a PSTN->VoIP gateway (with an E1 with multiple numbers pointing to it). I'd really like to be able to dial out to a SIP server like so: exten => _X.,1,Dial(SIP/${DNID}@hostname) I.e. the remote SIP server receives a SIP INVITE with a "To:" header containing the dialed number (e.g. 02085555555@computer.company.com). This is equivalent to having a

Interesting problem: An organization has departments. Each department has a single phone. Each department has multiple people. Each person within the organization has a direct dial incoming number. It's easy to set * up so that multiple DDIs get mapped to the same extension. What I'm wondering is if there's any way, with reasonably priced hardware, to notify the person who's

On 03/11/03 00:25, Mark Spencer wrote: > As a side note, I strongly would like to see someone implement a > client using libiax2 which implements IAX2 instead of the (now > obsolescent) IAX version 1. I'm implementing a Java-based IVR server (and yes, I know Asterisk does IVR, and no, it's not flexible enough to do what I want and no, it doesn't integrate well with the Java

I'm running an E400P. Every now and then Asterisk stops receiving incoming calls. This turns up in the messages log: Nov 25 10:49:12 WARNING[65541]: File chan_zap.c, Line 5793 (pri_dchannel): Ring requested on channel 1 already in use on span 1. Hanging up owner. Nov 25 10:49:15 WARNING[81926]: File chan_zap.c, Line 5793 (pri_dchannel): Ring requested on channel 1 already in use on

Hi. I'm trying to do some custom call logging, and I want to call an AGI script from a hangup handler to log call durations and things. Although the script executes, it isn't retrieving variables from the AGI interface. Looking closer, I realised the variables are actually getting unset before the h extension is reached. [foo] s,1,SetVar,foo=bar s,2,Play(audio/a-long-prompt)

We've changed E1 providers and I'm trying to reconfigure an E400P to make it work with the new lines. They're supposedly "standard" EuroISDN lines (in the UK). I'm initially just trying to get a single line up. I have the following in /etc/zaptel.conf: span=1,0,0,ccs,hdb3 bchan=1-15 dchan=16 bchan=17-31 loadzone=uk defaultzone=uk The LED on the back