similar to: IAX2 Ethereal Plugin initial release

Displaying 20 results from an estimated 3000 matches similar to: "IAX2 Ethereal Plugin initial release"

2003 Nov 27
5
IAX2 Ethereal plugin v0.3 is out
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
2003 Nov 18
3
Ethereal plugin for IAX2
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
2004 Jan 13
11
Best Linux Distribution
Hi my question is: which is the best distribution to work with asterisk? thanks mark
2003 Oct 13
2
e100p in norway?
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
2004 Jan 02
3
* Stresstool Help required
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
2003 Oct 14
1
outbound caller ID problem on PRI
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) >
2003 Sep 04
3
Call script after hangup
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:
2004 Jan 08
5
AbsoluteTimeout Users Messages
Hi, All Is there a provision for "AbsoluteTimeout" application to notify called and calling party of the reason why the call suddenly ended? This way, the parties will be much better informed, hence they will/should not think that their VOIP/telco provider(s) are providing bad service. Ta SJ
2003 Nov 07
2
Callgroups and Pickupgroups in Console/dsp
Hi all. I've made a patch for chan_oss.c to enable callgroups and pickupgroups in it (since wasn't enabled). I needed it for a special use of the console (pickup calls arriving to the console from another phone) btw, If someone is interested, I can submit a patch to the bugtracker. I won't do it until that's usefult for someone... since is a very special features that probably no
2003 Sep 11
1
UK Asterisk user, please pick up the white courtesy phone
So, I have submitted my configurations as public samples, and I should have expected this situation to arise. I changed all the relevant "private" configuration data in my samples to obfuscate or alter IP addresses, passwords, etc. However, I left my email address in voicemail.conf... Let me tell you, it took THREE messages sent by a distinctly British-sounding gentleman leaving
2003 Nov 10
1
Jitter Buffer on chan_sip
Hi, I would like to test chan_sip with a bigger jitter buffer. Does anybody know where in the code this is defined? I looked through it but could not find where. If anybody else can find it please let me know. Regards, Andres
2003 Nov 12
1
IAX needs a zaptel device?
Hi All, I'm currently running Asterisk with SIP phones and an ISDN card using chan_capi. I've just started to use IAX (GSM codec)over the Internet and the sound is adequate. However, there is an occasional 'glitch' in the audio resulting in lost sound or distortion. Is the distortion because I'm using zaprtc for timing instead of a zaptel card, or is more likely to be due to
2003 Nov 14
1
Re: 9. Zhone zplex (Angel Gomez Garcia)
Hi I have the last firmware for zplex, if you like i send it to you, about the second question 24s means 24 extensions so you can configurate as you wish as fxo or fxs. Att Yelson Vivas
2003 Nov 19
1
Play a "sound" after dialing a user...
I'd like to play a sound to a user I dial (via SIP) once they answer play the sound and then allow me to talk to them. The new Cisco 7960 SIP code allows to set lines to autoanswer via the speaker phone, I'd like to play a "tone" after it rings through and then talk... Any thoughts on how to do this?
2004 Jan 09
3
Very high delay
Hi I'm using a Teles ISDN passive card configured in modem.conf. when i make call from my sip client (xtex x-pro) to the external world i have more than 1 second of delay and echo very. There is some tuning to do? The performance is better with an active ISDN card or CAPI compatible driver? thanks mark balester
2004 Feb 02
1
Playing announcement to called user prior to Confirmation
Hello all, As I'm sure is pretty common, I have some extensions that dial mobile numbers after a local timeout. I would like to prompt the caller to record their name after the local timeout and have the recipient be able to hear the name prior to accepting the call. Recording the message is easy enough, so I thought about doing something like dumping them into MeetMe after they record
2004 Feb 03
2
busy tones
Hi When I call a phone with CAPI if the phone available I hear ringing ok but if the phone is busy I don't hear anything at all. Also, when I call a mobile phone and it is turned off I don't hear the operator voice answer me telling me that the request phone is turned off or unavailable. Any ideas? m
2004 Feb 03
1
GS and NAT
Hi all. Is it at all possible to have a GS B101 NATed with firmware 1.0.4.40? I've tried both STUN and not STUN. The odds seems best with stun because the phone registers with right ip adress. When the connection is made * sends rtp packets to the right destination AND port, but the phone doesn't accept the packets..... Should I burn my D-LINK 604 or upgrade the GS? /t
2004 Apr 23
1
CAPI and Extensions.conf Security problem
Hi, I've installing a AVM Fritz Card in my ASterisk Box I've configured everything and its running perfectly. The problem is that everybody is allow to call through it. Explaination: All users registered in Asterisk can make a call towards the ISDN network But, everybody from the Internet, knowing the extension of CAPI in the dialplan, can call through my Asterisk to any phone
2004 May 06
3
mpg123 versions ?
We find that mpg123 0.59r works best. mpg123 0.59s-mh4 = the devil. What versions does everyone use without problems. 0.59r is PERFECT bkw -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040506/937ae19c/attachment.htm