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Hello, I'm trying to migrate my system to FreeBSD and the Makefile for asterisk-addons fails in the first make clean: bash-2.05b# make clean "Makefile", line 56: Missing dependency operator "Makefile", line 57: Could not find .depend "Makefile", line 58: Need an operator make: fatal errors encountered -- cannot continue I would like to think there is no

...it for 1 year but never response me (error or OK). I need to do ctrl+c Any idea? Edgardo >>> jan@irial.com 06/21/05 10:07 AM >>> i know that there are extensive rework on the transfer in SIP at the moment. --On Tuesday, June 21, 2005 13:40:47 +0100 Victor Alvarez <victor@sentidocomun.es> wrote: > > Hi, > I'm afraid I don't know how to use the command Transfer. I have a couple > of SIP users in the system and although exten => 35,1,Dial(SIP/33) works > fine, exten => 35,1,Transfer(33) just don't work. All the description in > the wiki is...

Hi, Monitor and soxmix (m option) work fine in CVS Head, not in Asterisk 1.0.9, as the Wiki says. http://www.voip-info.org/tiki-index.php?page=Monitor+setup+sample Anyway I am wondering why asterisk 1.0.9 console shows on Hang up: "monitor executing ( nice -n 19 soxmix "//var/spool/asterisk/monitor/45/47-20050812-113631-in.wav"

Hello, My question is.. How does cdr_manager work? Does it suppose to populate cdr-csv/Master.csv? What about the cdr table on the database? What is the event some people talk about? I have changed (and reloaded) my configuration of cdr_manager.conf to ; ; Asterisk Call Management CDR ; [general] enabled = yes and it doesn't seem to make any difference. After originate a call from the

Hello, I can understand why asterisk is designed to not to allow two UAs with the same usr/pwd, http://lists.digium.com/pipermail/asterisk-users/2004-February/037284.html, but I have to find a solution for this. My first option is use SER as an extension end of Asterisk, to allow more than one SIP endpoint to register with the same details http://www.voip-info.org/wiki-Asterisk+at+large. I

Hello all! I am trying to load sip.conf from mysql database. I have followed the instructions at <http://www.voip-info.org/wiki-Asterisk+sip+mysql+peers>. Seems that the authentication (user & psw) works fine but I would like to get more information from mysql and I don't know how to retrieve it. Could anybody help me? Any idea about how to do it? Regards, Victor.

Hi, I'm trying to transfer an incoming call from the PSTN to another PSTN number through a SER - Asterisk system. SER doing only routing.. pstn call-> SER -> asterisk (call forward) -> SER -> pstn Logic for SER: If something comes from the pstn, send it to asterisk. If something comes from asterisk, send it to the pstn. Every time I am getting a "Got SIP response 481

...Regards, Dovid __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ------------------------------ Message: 15 Date: Thu, 2 Feb 2006 13:20:11 -0000 From: "Victor Alvarez" <victor@sentidocomun.es> Subject: [Asterisk-Users] Regarding cdr_manager.conf To: <asterisk-users@lists.digium.com> Message-ID: <013401c627fc$57348760$ed81a8c0@xana> Content-Type: text/plain; charset="iso-8859-1" Hello, My question is.. How does cdr_manager work? Does it suppose to populate c...

Hello, I would like to know how to add an iax user to the system, as simple as that, but it's hard to find one example. Just the entry to iax.conf and extensions.conf would be enough. I have already conect two asterisk servers, there are plenty of examples about how to do this but in this case I need to call an IAX extension in the same machine. I have no problem to create one IAX user and

Hello, I'm just wondering what is the situation today, 24 Nov 2004, regarding asterisk timer for freebsd. I would like to know if there is any way to run Meetme on Freebsd or if there is anybody currently working on it. Cheers, Victor. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, I'm trying to install CAPI Driver for Suse 9.2 and I found the documentation for this pretty old since It refers to Suse 8.2 ( http://www.voip-info.org/wiki-Asterisk+AVM+Fritz+CAPI+Driver+Install ). This is especially apparent when I look at the section of these instructions for altering "src.drv/makefile" to replace the occurance of "CARD_PATH". I tried to

Hi, Regarding capi debug, I don't know how to translate reasons like 0x3302 or infos like 0. I didn't find any 'translator' googleing capi debugging. Do you know about any 'translator' for this or should I be as clever as to know what a reason 0x3302 is? What is this debug for if I can't interpret it? Kind regards, Victor. >From capi debug: == CAPI Call

It's pretty obvious from the wiki that realtime and Nat don't befriend quite well. As It is obvious the necesity of both of them, mainly have clients under nat talking to an asterisk server. The question I would like to throw away is.. What would you do to have both of them? I have two possible solutions in mind. 1. Use static configuration for sip users with nat=yes. 2. Buy iax

Hi, I am 'playing' with asttapi which looks great on a first installation but I must be missing something regarding the source code because I haven't been able to work with it without problems. If you have played with this, you already know that the code to talk to Asterisk is placed in a file called asttapi.tsp which is in your windows/system32 folder. The thing here is, the

Hi, I'm trying to use the spandsp fax-back facility and despite I can send and receive faxes, this is not working fine. I would like to get a debug of what is going on. I am using the flag debug, but I don't know if txfax is generating any log information or where it is saving it. I just don't find anything. My line in extensions.conf is: exten =>

Hi, I have been playing around with the latest release of asttapi and I have found the 'hangup' problem already reported to the list here <http://lists.digium.com/pipermail/asterisk-users/2006-May/151260.html> Apparently hangup should be done by making use of UserEvent commands. So I have configured this context for being used when making calls from outlook: [outlook] exten =>

Hello everyone, I'm trying to find out a way to whisper the time remaining for a prepaid application on a established channel. Unfortunately I think there is a lack of PlayBack/Background commands which can be applied on a working channel as well as a lack of spy/whispering commands available via Asterisk Manager. Does anyone know how to implement this? Thanks a lot. Regards, Victor

Hello, I'm willing to implement call barring for incoming and outgoing calls and I would like to discuss it with the list first, since I think It can't be implemented in a 'natural way' and I will need to use agi scripting - database. Process would be: 1. incoming calls priority 1, call incoming.agi, select all the blocked cli's for the called user, if caller is on

Hi, How could I change the default permissions for voicemails? When I try to install the patch mentioned at http://www.voip-info.org/tiki-index.php?page=Asterisk+gui+vmail.cgi, I get the following response: patch < voicemail.patch patching file app_voicemail.c Hunk #1 FAILED at 39. Hunk #2 FAILED at 119. Hunk #3 FAILED at 296. Hunk #4 FAILED at 1248. Hunk #5 FAILED at 1273. Hunk #6