[asterisk-users] SIp Signalling

Is there a way to force asterisk to take care only of sip signaling without forcing it to take care of rtp traffic? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now:

[asterisk-users] Realtime SIP

Probably I did not read well the information I am concerning, if I am going to use ARA for the SIP and I have register => user:secret:[EMAIL PROTECTED]:port/extension how I should input that line? If I am going to delete it from the DB I am forced to reload everything or there is a way to tell ast

[asterisk-users] Asterisk 1.4.15 Voicemail

Hi after having installed asterisk 1.4.15 my voicemail does not work anymore. Am I the only one? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.d

Re: [asterisk-users] [Fwd: Re: VoiceMail hangup]

l extension. > - I hear "you have 1 new message. Press 1..." 1 pressed (without waiting for > the message playing) > - Asterisk hangups. > > I'm not always able to replicate the problem but, as "Il Neofita", I'm using > the italian prompts... could be a

Re: [asterisk-users] VoiceMail hangup

Probably I was not able to explain myself properly however, for some measge this what happen -- Playing '/var/spool/asterisk/voicemail/default/300/Old/msg0003' (language 'it') == Spawn extension (servizi, , 1) exited non-zero on 'Local/[EMAIL PROTECTED],2' I cannot listen the message a

Re: [asterisk-users] VoiceMail hangup

07, 15:14 -0500 schrieb Il Neofita: > > > Hi > > additional information if I am going to wait at least 3 seconds after > > the voicemail starts to give me the instruction I am able to listen my > > messages. > > But why I need to wait? > > > > On Nov

Re: [asterisk-users] VoiceMail hangup

Hi additional information if I am going to wait at least 3 seconds after the voicemail starts to give me the instruction I am able to listen my messages. But why I need to wait? On Nov 12, 2007 2:28 PM, Il Neofita <[EMAIL PROTECTED]> wrote: > Hi, > with some messages the voicemailmai

[asterisk-users] VoiceMail hangup

Hi, with some messages the voicemailmain after give me the information about the call (Days, hours and minutes) it finish. Whant can I check for solve this problem? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users

Re: [asterisk-users] Help with loop counting?

Hi I believe that exten => s,7,GotoIf($[${trips}=4]?,8) the , should be : On 10/24/07, Phil Knighton <[EMAIL PROTECTED]> wrote: > > Hi > > I have a situation where I want to be able to count how many times a > caller goes round a loop of "Please hold...", "please continue to hold". I > have

[asterisk-users] Force codec order

There is a way to force the order of the codecs in the sip.conf since the allow seams to let know only the accepted codec. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update option

[asterisk-users] Strange behaviour afetr update from 1.2 to 1.4

Hi, I update from asterisk 1.2 to 1.4 and I have some problems. In the extensions I used DIAL(SIP/100&SIP/101,30,tTr) if I receive a call from an external providers now in 1.4 I recieve only one ring What can I do to solve this problem? ___ --Bandwidth an

Re: [asterisk-users] GTALK problem

Thank you I need to wait the international version of gtalk On 10/11/07, Philippe Sultan <[EMAIL PROTECTED]> wrote: > > > If I calling asterisk with GTALK in english everything is ok, however, > some > > of my friends with the italian version of gtalk they cannot have the > audio. > > Audio probl

[asterisk-users] GTALK problem

Hi, I installed gtalk on asterisk 1.4.12.1, I change on rtp.conf the port from 1000 to 4 If I calling asterisk with GTALK in english everything is ok, however, some of my friends with the italian version of gtalk they cannot have the audio. Is it a bug? Or I did some mistake __

[asterisk-users] Curiosity Max Calls

Hi is there a tool to know what was the maximum calls that asterisk managed? Thank you ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com

[asterisk-users] Call hangup after 60seconds

Hi, I have a client (xlite) connected to my server, on the server I have type=friend and siptimeout=60, canreinvite=yes and dial with tT option, the server is listening on port 5060. However, xlite is connect to a router where the port 5060 is blocked, therefore, I am using 5065 and I have an iptab

Re: [asterisk-users] Strange Behaviour

Thank you I will try tonight On 9/10/07, Anselm Martin Hoffmeister <[EMAIL PROTECTED]> wrote: > > Am Montag, den 10.09.2007, 05:14 +0200 schrieb Il Neofita: > > On 9/9/07, Anselm Martin Hoffmeister <[EMAIL PROTECTED]> > > wrote: > > Am Sonntag, de

Re: [asterisk-users] Strange Behaviour

On 9/9/07, Anselm Martin Hoffmeister <[EMAIL PROTECTED]> wrote: > > Am Sonntag, den 09.09.2007, 20:16 +0200 schrieb Il Neofita: > > Well, it seems there are differences between those accounts then. > > You might want to post your sip.conf, and -if that is possible- the ATA

[asterisk-users] Strange Behaviour

Hi, my ATA has two phones attached and the possibility to set different accounts. I put two account of my asterisk server, however, it is able to call only with the second one in order to the sip.conf and the first it gives me 403. And idea how to solve it? _

[asterisk-users] extra field

Hi, I am using my asterisk server like a gateway and one provider ask me to pass an extra field with the IP of the peer that is using the connection, probably to have more control on the authentication. I was wondering how I can implement this. Thank you __

Re: [asterisk-users] TDM400p Loaded only once

AIL PROTECTED]> wrote: On Thu, Mar 01, 2007 at 06:21:38AM -0500, Il Neofita wrote: > Hi > when I turn on my PC I able to load the drivers and start my card, > if I reboot the PC I have the following error > > ztcfg -vvv > > Zaptel Configuration > == > >

[asterisk-users] TDM400p Loaded only once

Hi when I turn on my PC I able to load the drivers and start my card, if I reboot the PC I have the following error ztcfg -vvv Zaptel Configuration == Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) Channel 02: FXS Kewlstart (Default) (Slaves: 02) Channel 03:

[asterisk-users] Strange Noise

Hi, today with my asterisk during a call I had a very strange noise, it was the typical noise that you can have when your device uses a bad power supply. I change phone and I had the same behavior after I while I tried again and the noise was disappeared. What can I check? __

Re: [asterisk-users] Strange behaviour with Dial cmd

Yes, but I would like to try a number and after to try a second one. Any Idea how to avoid this. On 2/18/07, Eric ManxPower Wieling <[EMAIL PROTECTED]> wrote: C F wrote: > Asterisk supports this directly by issuing the hangup command before > the answer command. However, when using an analog i

Re: [asterisk-users] Strange behaviour with Dial cmd

Ok thank you a lot!!! On 2/15/07, Yuan LIU <[EMAIL PROTECTED]> wrote: >From: "Il Neofita" <[EMAIL PROTECTED]> >Date: Thu, 15 Feb 2007 03:37:14 -0500 > >But I tought that hangup was suppose to close the call, however, is not the >case and a really di

Re: [asterisk-users] Strange behaviour with Dial cmd

On 2/14/07, Yuan LIU <[EMAIL PROTECTED]> wrote: Well, you'll have to decide how you want to "hang up" the caller: Do you want him/her to be ignored, or to be told that you are not available (like an answering machine)? You also need to tell Asterisk how to determine if the next "invite" comes

Re: [asterisk-users] Strange behaviour with Dial cmd

. On 2/14/07, Yuan LIU <[EMAIL PROTECTED]> wrote: >From: "Il Neofita" <[EMAIL PROTECTED]> >Date: Wed, 14 Feb 2007 19:30:51 -0500 > >I have this simple context > >I am register to an external provider and when I am not home I would like >to transfer the ph

[asterisk-users] Strange behaviour with Dial cmd

I have this simple context I am register to an external provider and when I am not home I would like to transfer the phone outside The problem that the call goes in loop I cannot understand why. Can you figure out my error? Thank you sip.conf register => user:[EMAIL PROTECTED]/400 [inside]

Re: [asterisk-users] asterisk 1.4 FC5 and Gtalk

Sei riuscito? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] RE: asterisk 1.4 FC5 and Gtalk

On 2/10/07, Patrick <[EMAIL PROTECTED]> wrote: Where can I find that patch? Thanks, Patrick Hi Patrick, I downloaded the patch from here http://bugs.digium.com/view.php?id=7764 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-u

Re: [asterisk-users] RE: asterisk 1.4 FC5 and Gtalk

Hi, I tried thousands of time and finally I am a step closer to the solution. I recompile iksemel with the option --prefix=/usr I erase my zaptel-1.4, asterisk-1.4 and asterisk-addons-1.4, re-extracting everything from the tar recompile everything and now jabber is working or almost. When I receiv

[asterisk-users] asterisk 1.4 FC5 and Gtalk

JABBER: gtalk_account OUTGOING: localhost*CLI> jabber show tes JABBER: gtalk_account INCOMING: http://etherx.jabber.org/streams"; xmlns="jabber:client"> [Feb 9 21:11:15] ERROR[2061]: res_jabber.c:482 aji_act_hook: gnuTLS not installed. I installed all the gnutls but I still have this error [E

[asterisk-users] Re: Chan_Cellphone

I start the patch and automatically created the file. But now on the menu I cannot select chan_cellphone I launched ./bootstrap.sh and after ./configure in my /usr/include/bluetooth I have the header but I cannot select the option any idea? On 2/9/07, Il Neofita <[EMAIL PROTECTED]> wrote

Re: [asterisk-users] BindPort

The point is to use more than one port, I think the only way is to use the redirect from iptables On 2/6/07, Giorgio Incantalupo <[EMAIL PROTECTED]> wrote: Ciao, just change port value in sip.conf. Giorgio Il Neofita wrote: > Hi, > I was wondering if it is possible to set aster

[asterisk-users] Chan_Cellphone

Hi, I download the last svn and I also look around but I cannot find the source, I only found the patch http://bugs.digium.com/print_bug_page.php?bug_id=8919 any one can help me out. thx ___ --Bandwidth and Colocation provided by Easynews.com -- aster

[asterisk-users] BindPort

Hi, I was wondering if it is possible to set asterisk in order to listen to different ports for the sip or I need to do this operation with iptables? All of this since some time the port 5060 is blocked. Thank you ___ --Bandwidth and Colocation provided

Re: [asterisk-users] chanskype

Hi, I was wondering if someone had problems with chanskype. Since I am wondering if they are a credible company or not. See you On 1/20/07, Moises Silva <[EMAIL PROTECTED]> wrote: > Hi, > I tried the try version of chanskype, however, everytime that I make a call > asterisk generate an error S

[asterisk-users] chanskype

Hi, I tried the try version of chanskype, however, everytime that I make a call asterisk generate an error Anyone has experience with this? Since I tried to contact the support but they never replied. Thank you ___ --Bandwidth and Colocation provided b

Re: [asterisk-users] Looking for toll free in Italy

I do not think that there are some company that offer a toll free number (Numero verde in italian) But contact on of these three providers http://www.eutelia.it/tlc/ http://www.unidata.it/ http://messagenet.it/ If they have one of these should be able to give to you See you On 1/8/07, CM Rahman

[asterisk-users] Strange error

Someone know why my asterisk gives me the following msgs? Thank you - Got SIP response 603 "Declined (no dialog)" back from X.X.X.X -- Got SIP response 603 "Declined (no dialog)" back from X.X.X.Y -- Got

Re: [Asterisk-Users] WebPhone

On 7/3/06, Jean-Denis Girard <[EMAIL PROTECTED]> wrote: But I'm not sure that MozPhone is what the original poster asked.No, however, I like to read all these different point of view. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Us

[Asterisk-Users] WebPhone

Hi,someone know a good webphone, possibily a free oneThx ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Canreinvite

es going on. pFrom: "Il Neofita" < [EMAIL PROTECTED]>To: "Asterisk Users Mailing List - Non-Commercial Discussion" < asterisk-users@lists.digium.com>Date: Sun, 18 Jun 2006 05:01:20 -0400Subject: Re: [Asterisk-Users] Canreinvite This is the dial in extensionsext

Re: [Asterisk-Users] Canreinvite

cosa vedo a console    -- Executing Dial("SIP/40001-3760", "SIP/40002|30") in new stack    -- Called 40002    -- SIP/40002-4753 is ringing    -- SIP/40002-4753 answered SIP/40001-3760     -- Attempting native bridge of SIP/40001-3760 and SIP/40002-4753srvlinux*CLI> sip show channelsPeer

Re: [Asterisk-Users] Canreinvite

This is the dial in extensionsexten => _40001,1,Dial(SIP/40001,30)    exten => _40002,1,Dial(SIP/40002,30)    ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.

[Asterisk-Users] Canreinvite

I put canreinvite=yes in my sip, for a sipura 3000 and a xlite, however, if I call the traffic still go throw the asterisk. How come? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options

[Asterisk-Users] Zaptel Module.symvers missing

Hi,when I am going to compile the zaptel I receive this messageake -C /lib/modules/2.6.12-12mdk/build SUBDIRS=/usr/src/asterisk/zaptel-1.2.5 XPPMOD= modulesmake[1]: Entering directory `/usr/src/linux-2.6.12-12mdk '  WARNING: Symbol version dump /usr/src/linux-2.6.12-12mdk/Module.symvers   i

Re: [Asterisk-Users] NOTIFY Problem

I agree with you, but I would like to find a way to use the notification. I tough that there was a work around.On 4/29/06, tom < [EMAIL PROTECTED]> wrote:Il Neofita wrote:> Hi,> one of my WiFI phone has problem with the notify asterisk signal to me > the following> Apr 29 06:49:16 WARNING[6455] cha

[Asterisk-Users] NOTIFY Problem

Hi,one of my WiFI phone has problem with the notify asterisk signal to me the following Apr 29 06:49:16 WARNING[6455] chan_sip.c: Host '192.168.100.124' does not implement 'NOTIFY' In theory the phone support this function.Any idea? ___ --Bandwidth and Co

[Asterisk-Users] Remote UNIX connection disconnected over and over

Hi,I am pretty sure that you already answer to this question, but I was not able to find the solutionon the console I have over and over the following msgs -- Remote UNIX connection disconnected    -- Remote UNIX connection disconnected     -- Remote UNIX connection disconnected    -- Remote UNIX c

[Asterisk-Users] How to use the cmd SMS

Hi,I try to use my phone that has a SMS capability with asterisk.I am not able to receive SMS, someone can help me out?This what I am able to have but nothing more -- Executing SMS("SIP/503-7d2e", "508|sa") in new stack     -- SMS TX 93 00 6D ___ --Bandwi

[Asterisk-Users] chan_gsm_bt Impression

Hi,Has anyone proved the chan_gsm_bt ??Any impression? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Error Header field Via

Someone know the meaing of this error? chan_sip.c:3853 copy_via_headers: No header field 'Via' present to copy ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.

[Asterisk-Users] Codec Problem

I have the license for G729, however I need to use a different codec for the prepaid service, but when the call is started I have this error Asked to transmit frame type 256, while native formats is 4 (read/write = 4/4) ___ --Bandwidth and Colocation prov

Re: [Asterisk-Users] H323 on way voice

This is my debugtried with netmeeting I can still hear but when I talk nothing happenmygw*CLI> h.323 debugH323 debug enabledmygw== New H.323 Connection created.    -- Setting up Call    -- oCall token:  [ip$192.168.1.219:1057/8226]     -- oCalling party name:  [myPersonal]    -- oCalling party numb

[Asterisk-Users] H323 on way voice

Hi,I installed H323, however when I make a call from SIP Phone -> Asterisk H323 -> Provider H323 the provider can hear me, but I cannot hear nothing.The asterisk is 1.2.6 with G729 license, and the asterisk is connect direct to internet with a public IP. Any thoughts? __

[Asterisk-Users] channel.c:787 channel_find_locked: Avoided initial deadlock for '0x8446b50', 10 retries!

I never so this error.I am using H323 with Asterisk 1.2.6 Any idea what can be the problem? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/

[Asterisk-Users] H323 behind a Firewall

There is a proble to put an H323 Asterisk server behind an iptables firewall? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/aster

[Asterisk-Users] H323 Info

Hi,I compiled for my asterisk 1.2.4 the openh323but when I give this commandh.323 show codecsI receive thisAllowed Codecs: Table: Set:I cannot test with msn if everything is working since I am outside and I cannot access to the firewall. Someone can tell me if I need to install the oh323 si

Re: [Asterisk-Users] TDM400P busy

Yes, I changed.Thank youOn 3/27/06, Infobox Peru <[EMAIL PROTECTED]> wrote: Maybe your lines use polarity reversals for hangup detection.On 3/27/06, Il Neofita <[EMAIL PROTECTED] > wrote: Hi,I have a TDM400P with 4 FXO The TDM after it receive a call do not hangup properly, it ta

[Asterisk-Users] TDM400P busy

Hi,I have a TDM400P with 4 FXO The TDM after it receive a call do not hangup properly, it takes the line occupied. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://li

[Asterisk-Users] AstCC

Hi,I am wondering if it is possible with astcc to make a second call without hangup and be oblige to re-enter all the codes.Any idea how to do?Thank you ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE

[Asterisk-Users] MWI problem

Sorry if I am always here asking for MWI, but I do not know how to solve this issue, I have my ATAs (Azatel 200 and Fritz!Box) that they think that I have a message waiting. Anyone knows how to solve this issue? Thank you ___ --Bandwidth and Colocation p

[Asterisk-Users] WMI Problem

Hi, I was not able to find any indication for this problem that I have right now. My phones connected to an Azatel 200 they always indicate that I have a message waiting to be listen. However, I do not have any message. I also checked using the console "show voicemail user for context" but I have 0

[Asterisk-Users] ATA does not register

I am not able to register an  external ATA on my asterisk 2.0 Beta This is the debug Any idea? server01*CLI> <-- SIP read from CLIENTIP:5060: REGISTER sip:SIPSERVERIP SIP/2.0 Via: SIP/2.0/UDP PRIVATEIP;rport;branch=z9hG4bK455E4AEA9A9954FB135D6D788DA2 From: ;tag=1564789518 To: Call-ID: [EMAIL PRO

[Asterisk-Users] WMI problem

I installed astersik 1.2beta and from that point the led that indicate a new call flash. The ATA installed is an AZATEL. Any idea what can I check? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lis

[Asterisk-Users] TXFAX

Hi, do you know what it means the following: "Call failed to go through, reason 3" I received it when I try to send a FAX and no one answer it. Thank you ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asteris

Re: [Asterisk-Users] Txfax

This is the script I do not remeber the name of the author For me is working On 9/9/05, Chris Shipman <[EMAIL PROTECTED]> wrote: Thanks Luki,Maybe it was the wrong place to mention it.Back to the other matter: The program will create a PDF or tif. Itcan submit the fax by FTP or by AstF

Re: [Asterisk-Users] sending fax

HI Chris I am interested, I would like to know how I can have the opportunity to test your program. On 9/9/05, Tzafrir Cohen <[EMAIL PROTECTED]> wrote: On Thu, Sep 08, 2005 at 06:42:49PM -0500, Chris Shipman wrote:> I started working on a program using Ghostscript and Redmon to generate> the ti

Re: [Asterisk-Users] sending fax

The site is in italian, and you need to register your self in order to download the script. The script is in perl and you need to start in order to simulate the hylafax daemon after that you can use WHFC On 9/8/05, Matthew Gibson <[EMAIL PROTECTED]> wrote: Sorry to interrupt :)But I believe what yo

[Asterisk-Users] Txfax

Is there some way to know if the fax was received correctly or not? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSC

[Asterisk-Users] Extension a

Hi, I would like to use the * when I am in the asnwer machine, but I received a message asking for the temporary pass code. Where I need to put this pass? I am using asterisk 1.2.0 beta 1 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Aster

Re: [Asterisk-Users] Asterisk architecture

The call generate from branch2 can be send to the asterisk in Branch1 with a trunk the same think the call received from branch1 the only thing that is not cleat  how you want transfer automatically the call received from the pstn. What rule you want use?On 9/5/05, housi mueller <[EMAIL PROTECTED]>

Re: [Asterisk-Users] Asterisk architecture

>From my point of view I do not see any issue with that scenario. On 9/5/05, housi mueller <[EMAIL PROTECTED]> wrote: I am new with asterisk and hope somebody can help me.   Is a configuration like shown on the picture with asterisk correct?  Some phone calls arriving in Branch 1 should be redirec

Re: [Asterisk-Users] sending fax

Hi, I found on a forum a script that "emulate a hylafax" this is the linkhttp://www.vocesuip.com/viewtopic.php?p=2423 You can use the WHFC in order to send a fax to asterisk. On 9/5/05, Harald Klein <[EMAIL PROTECTED]> wrote: Hi Chris, Hi Arne,Am 5.9.2005 schrieb "Chris Shipman" <[EMAIL PROTECTED]

Re: [Asterisk-Users] Problem with include

Sorry, I'm using the 1.0.9On 9/1/05, Kevin P. Fleming <[EMAIL PROTECTED]> wrote: Il Neofita wrote:> Hi,> I put on sip.conf the following line>> #include "sip.d/*.conf"You neglected to include the most important piece of information: what

[Asterisk-Users] Problem with include

Hi, I put on sip.conf the following line #include "sip.d/*.conf" inside I have files like that provider1.conf provider2.conf But asterisk does not want to load it This is the error Sep  1 13:18:35 VERBOSE[8756]:   == Parsing '/etc/asterisk/sip.d/*.conf': Sep  1 13:18:35 VERBOSE[8756]:   == Par

[Asterisk-Users] TXFAX() status

Hi, I'm using a script in order to send out my faxes with the application txfax, therefore, I do not know how to see if the faxes are sent. Any idea? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@

Re: [Asterisk-Users] ASTCC and cdrs

Thank you was that the problem. On 8/25/05, Darren Wiebe <[EMAIL PROTECTED]> wrote: > When did you install it? Try running the "update database" function > from the configure menu. > > Darren Wiebe > [EMAIL PROTECTED] > > Il Neofita wrote: > > &g

[Asterisk-Users] ASTCC and cdrs

My installation of ASTCC does not update the cdrs tables . It is a problem of ASTCC or it is a configuration problem? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or

[Asterisk-Users] Calling Card Application

I am looking for a calling card application which is able to advise me during a call when the credit is almost finish. For examples 1 minute before the end of the credit. Thank you ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://li

[Asterisk-Users] X100P compatible

Why my X100P detect the ring after 3 o 4 seconds? The funny thing that when I have an incoming call asterisk receive a signal but the commands start after 3 or 4 seconds. Moreover, when the call end the hungup has the same delay. any ideas? ___ Asterisk-U

Re: [Asterisk-Users] initiating Monitor during call

I put these lines on features.conf in asterisk CVS-v1-0-08/16/05 [featuremap] blindxfer=> ## automon => *1 atxfer => *2 and I also added the options wW in the dial, howerver, when I call and I press *1 nothing happen. Any ideas? On 8/18/05, Matt Florell <[EMAIL PROTECTED]> wrote: > Yes, there a

Re: [Asterisk-Users] TAFM

Hi, I checked but I did not find any info regarding the config files. I tough that I configured everything in the right way but I am not able to see anything on the web page. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.dig

[Asterisk-Users] TAFM

Hi, I installed this program but I am not able to configure, it does not want to work. Someone can help me? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update op