similar to: Re: Asterisk-Users digest, Vol 1 #237 - 11 msgs

Hello, I'd like to ask if there are any news about CE certification of the E1 boards. I know that the T1 boards are FCC certified but I'd also like to know what is the status for CE certification. Thanks for any input, Vlasis Hatzistavrou.

Hello, Does anyone know any solution to this? Or is Asterisk-Dev a more suitable list to ask this question? Best regards, Vlasis Hatzistavrou. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Vlasis Hatzistavrou Sent: Thursday, February 16, 2006 3:43 PM To: asterisk-users@lists.digium.com Cc: 'Vlasis

Hello, On an Asterisk 1.4.33.1 in a simple scenario: [test] exten => _X.,1,Dial(SIP/12345 at peer01,,,) exten => i,1,Hangup(${HANGUPCAUSE}) exten => t,1,Hangup(${HANGUPCAUSE}) exten => h,1,Hangup(${HANGUPCAUSE}) I have noticed that no matter what value we set in the Hangup(<cause code>) commands, if the call is not answered by peer01 for any reason, the actual cause code

Hello all, We have been testing Asterisk RC2 with the native H323 channel driver. We followed the instructions with the needed OpenH323 and PWLib versions and everything compiled ok. Operation of the driver seems ok, except from 2 main points: 1) Audio is passed between the two ends of the call only after the call is answered. This was not the case with previous versions of Asterisk (0.9.2

Klaus-Peter (while hardly able to see through my watery eyes), You've earned yourself a (potentially rackmountable) 19" Dutch cheese as soon as the zaptel driver is finished and tested. In fact, I don't even have to mail it, I can pratically bring it over ! Although I'll be monitoring your site on a 24hrs/day basis from now on, please keep me posted on both the driver and the 4

asterisk-users-request@lists.digium.com wrote: >Send Asterisk-Users mailing list submissions to > asterisk-users@lists.digium.com > >To subscribe or unsubscribe via the World Wide Web, visit > http://lists.digium.com/mailman/listinfo/asterisk-users >or, via email, send a message with subject or body 'help' to > asterisk-users-request@lists.digium.com > >You can

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Hello, I just tried to compile OH323 0.6.3b on a RH9 machine with Asterisk 1.0 RC2 installed but failed. I applied the patch to the required OpenH323 library according to the instructions, and set the proper directories in the Makefile. Here is what I receive after I issue make: ******************************* g++ -DP_USE_PRAGMA -fno-rtti -ffunction-sections -fdata-sections -D_REENTRANT -

Hello. The procedure so that it works you can find in: http://www.voip-info.org/wiki-Convert+WAV+audio+files+for+use+in+Asteris k a the files .wav chmod 755 file.wav sox file.wav -r 8000 file.gsm resample -ql chmod 755 file.gsm in extensions.conf xxxx=> xxx,x,playback(file) Ing Javier Rios Ing de Proyectos 04167285748 212 2637246 /2637187 -----Original Message----- From:

Hello, I have used the Dial option 'r' before in older Asterisk versions and it behaved as expected, that is, Asterisk would always give ringback audio before the call was answered no matter what. It has been a while that I have used version 1.4.33.1 without any the 'r' option. Now that I had to use it for a while, I noticed that 'r' would not give ANY audio until the

Hello all, Sorry for the slightly off-topic issue, I need to have a capture from a network sniffer (like Ethereal for example) from a call setup with the MGCP protocol. I thought that since Asterisk now supports MGCP some of the people who develop the MGCP channel driver may have such a capture available. I need this for my MSc thesis and unfortunately, I don't have any MGCP compliant

Hello, I would like to ask if anyone has solved the problem with Asterisk+ISDN4Linux cards, where there are no call progress tones or announcements from the PSTN when we dial ouot through the i4l card. For the moment, if we don't inject the r option in the Dial command, there is only silence during the call negotiation... Using Asterisk RC2 with Eicon passive PCI 2.01 card... Thanks for

Hello, I am trying to write an AGI in Perl and I need to execute a function upon answer of a call. I know that there is the possibility to use the M() option in the Dial command in order to do what I need, but that would mean that I would have to incorporate the function's work in a different AGI program, and I need to avoid this. So, I would like to know if such an option is available in

Kapejod is working on a fix for the CDR problem in bristuff. See below ----- Forwarded message from kpj@junghanns.net ----- Resent-From: tzafrir.cohen@xorcom.com Resent-Date: Fri, 10 Feb 2006 13:01:25 +0200 Resent-Message-ID: <20060210110125.GU16880@xorcom.com> Resent-To: tzafrir@cohens.org.il Envelope-to: tzafrir.cohen@xorcom.com Delivery-date: Fri, 10 Feb 2006 05:19:50 -0500 Date: Fri,

I don't know if 911 uses caller ID or BTN (Billing Telephone Number) 900 calls, operator calls, and 800 calls use the BTN not the Caller ID... Anyone???? 3. Re: E911 and asterisk (Martin Pycko) Message: 3 Date: Mon, 21 Jul 2003 12:05:38 -0500 (CDT) From: Martin Pycko <martinp@digium.com> To: <asterisk-users@lists.digium.com> Subject: Re: [Asterisk-Users] E911 and asterisk

Hello. The procedure so that it works you can find in: http://www.voip-info.org/wiki-Convert+WAV+audio+files+for+use+in+Asteris k a the files .wav chmod 755 file.wav sox file.wav -r 8000 file.gsm resample -ql chmod 755 file.gsm in extensions.conf xxxx=> xxx,x,playback(file) Ing Javier Rios Ing de Proyectos 04167285748 212 2637246 /2637187 -----Original Message----- From:

asterisk-users-request@lists.digium.com wrote: >Send Asterisk-Users mailing list submissions to > asterisk-users@lists.digium.com > >To subscribe or unsubscribe via the World Wide Web, visit > http://lists.digium.com/mailman/listinfo/asterisk-users >or, via email, send a message with subject or body 'help' to > asterisk-users-request@lists.digium.com > >You can

Dear asterisk users, I am using Asterisk and I a m a new user. Before it was working properly. Since two days, users can not get registered : users registered timeout. Those are the results of commands 1. /var/log/asterisk#asterisk-rvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvv == Parsing '/etc/asterisk/extconfig.conf': Found Asterisk 1.2.1, Copyright (C) 1999 - 2005 Digium. Written by

right, so far, here is what I've done: I succeed in take in a new variable the UUS1 field sent with the connection request for incoming calls. It was quite simple afterall... (I just had to find where the data CMSG->Useruserdata is coming in chan_capi.c) Now I would like to know where this field is instanciated for outgoing calls in order to control this step? I am looking for that but I

Hi all, I could use MP3Player to play local sound (e.g: /usr/sound/abc.mp3) but I could not use it to run a remote stream, if I use mpg123 in command line, I can hear the audio ( /usr/bin/mpg123 http://...), but the same remote mp3 file could not be replay with asterisk. I would appreciate with any suggestion. Phuong Here are the log message: -- Starting simple switch on 'Zap/3-1'