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...0500 > Resent-From: tzafrir@gadot.org.il > Resent-Date: Fri, 10 Feb 2006 11:32:00 +0200 > Resent-Message-ID: <20060210093200.GG31420@gadot.org.il> > Resent-To: tzafrir@xorcom.com > Subject: RE: [Asterisk-Users] Corrupt CDR records in Asterisk 1.2.x To: > asterisk-users@lists.digium.com > From: asterisk@frameweb.it > Date: Fri, 10 Feb 2006 10:22:19 +0100 > Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users@lists.digium.com> > > Yes, everybody of us use zaphfc. > No problem at all with zap channel that I have installe...

Another trivial question: Is there a "place" where all the parameters are documented ? In example (my example!) I would like to know the meaning of a lot of parameter that can be used in sip.conf, A lot of these keywords are intuitive keywords (i.e. NAT=YES/NO;PORT=5060; context=xxxxx) but other are not (at least for me) i.e.: type = peer, friend insecure=very host=dynamic and so on.

...steris 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-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of asterisk-users-request@lists.digium.com Sent: jueves, 06 de noviembre de 2003 12:10 To: asterisk-users@lists.digium.com Subject: Asterisk-Users digest, Vol 1 #1808 - 13 msgs Send Asterisk-Users mailing list submissions to asterisk-...

Hi all, I just arrived in Italy from Cebit, qhere I spoke with digium and Beronet people. They told me to try to use the mISDN stack to drive beronet and the new upcoming digium ISDN Cards. SO I searched, find http://www.beronet.com/download/card_installation_guide.pdf, and I immediately got the error: asterisk01:~ # cd /usr/src/install-misdn/ asterisk01:/usr/src...

Is there any way to customize VoiceMail ? I would like to customize the message played to callers sent to the voicemail becouse the extension is busy or otherwise unavailable. Is it a way to record a welcome message and use it ? thanks in advance, Andrea Chi ricevesse questa mail per errore e' gentilmente pregato di cancellarla. Visitate il sito http://www.frameweb.it

I realize I cannot use inband audio for phones (voicemail and internal ivr, password for external trunks and other thing not working) So I put everywhere rfc2833. Doing this, anyway, make any EXTERNAL IVR NOT working. I see a lot of posts about this, but no solution, becouse using inband audio (which works for outside...) breaks inside IVR Is it possible to define to use inband audio ONLY on

I have a problem with CDR recording in Asterisk 1.2.x. This is the situation: An Asterisk 1.2.0 (BRIstuffed-0.3.0-PRE-1 with florz) machine with a single HFC-S ISDN BRI card. I log the call records to both the Master.csv and MySQL. The problem is that when an incoming call from the ISDN line is logged to the CDR, the "src" and the "clid" field show up as something like

An example for Radius is calling cards.. I can use * for this kind of service... With platforms that use Radius Server. -----Mensaje original----- De: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] En nombre de asterisk-users-request@lists.digium.com Enviado el: Lunes, 17 de Noviembre de 2003 07:16 p.m. Para: asterisk-users@lists.digium.com Asunto: Asterisk-Users digest, Vol 1 #1918 - 9 msgs Send Asterisk-Users mailing list submissions to...

...em.so=no chan_capi.so=yes But in capi.conf i really don't know what exactly to put, i left it as it comes, but i don't know how to set this file up. Any one is a chan_capi guru???? Regards Diego -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040217/23eeae35/attachment.htm

hi all, I am trunking via iax2 2 asterisk serverses if both of them have static ip addresses, I can connect them using no password, password or auth rsa with a pair of keys. If one of them has dynamic ip address and need to register on the other server, I can connect them with no password, but I am not able to do that using keys. The question is: which is the right register syntax to use when

I am having a strange problem. On one * box I setup the fax recive, via spandsp -app_rxfax I have no problem here. On a second box I did the same. The resulting PDF appear "corrupt". If I transmit the same fax to both * box, the tiff files received are the same. A deeper analysis shows the only problem is the width and heigth of the document In the first PDF, I see

It seems to never fail - after 3 to 5 minutes SIP -> IAX calls drop audio on one side. I place a call out through voipjet, and call quality is flawless. However a few minutes later the person who I'm talking to can no longer hear me. I can still hear them. What should I look for to resolve this? Has anyone else had this problem? Using last night's CVS this problem still exists.

Does anyone have a document I can use as a guide on how to load a SIP image on a cisco 7960 phone? Ryan

If you define a sip peer, wheather or not you put an entry in the field OUTBOUND CID, if you dial an external extension (let's say an extension on another asterisk server, connected via IAX2 connection) the callerid received by the foreign asterisk is device <YOURNUMBER>: i.e device <567> If you take a look at etc/asterisk/sip_additional.conf, you can see under the SIP extension

...creased file count. Not sure exactly what this refers to but can someone point me in the right direction? Or am I on the wrong track? Thanks. JR JR Richardson Engineering for the Masses -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070321/6bcadd6e/attachment.htm

...eedom.org> To Sent by: Asterisk Users Mailing List - asterisk-users-bo Non-Commercial Discussion unces@lists.digiu <asterisk-users@lists.digium.com> m.com cc Subject 01/06/2...

I'm pleased to announce that Flash Operator Panel 0.26 has been released! FOP is a GPL'd switchboard type application for the Asterisk PBX. It runs on a web browser with the flash plugin. It is able to display information about your Asterisk box in real time. It is included in FreePBX, Asterisk@Home, DeStar, startShop, and several other projects both free and commercial. You can grab the

...OGOUT" We are monitoring the database and it seems like the application doesn't connect to it. Does anybody in this have made this work? Can someone help me please?? Thanks, Robson -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050531/fc7401dd/attachment.htm

I am trying to use misdn insted of zaphfc to drive two billion isdn cards zaphfc is ok, but the problem with cdr and the fact tha you always have to wait the bristuffed version of asterisk took me to try another way. so I downloaded the misdn installation script from beronet for the last version ( I am using asterisk stable 1.2, so now is 1.2.5) wget

Hi all as per the new release notice for 13.33.0 received today - can anyone advise me the max limit of the string to the Dial Command - see * [ASTERISK-27946 <BLOCKED::https://issues.asterisk.org/jira/browse/ASTERISK-27946> ] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION when it shouldn't I have been fighting with this issue for months trying to find a solution I