similar to: Calls die when the answering party transfers

If your phones are setup to connect to the asterisk box by name, then a smart DNS server can just point phones to the backup box after failure. However, since asterisk running on the backup box doesn't know about the phones, this is only half the solution ________________________________ From: Mohamed A. Gombolaty [mailto:mgombolaty@noorgroup.net] Sent: Thursday, June 30, 2005 8:30 AM To:

Dear All, I have downloaded the xlite version 2.0 for windows and I made the following conf in the xlite itself as the document suggested in order to make it work with Asterisk but still it doesn't work as a matter of fact when I tried to make a tcp dump I can see no packets going between the windows client and the Asterisk server at all, here is the my conf on the xlite itself: in the

Hi, I'm looking for a reliable provider to use mainly for outgoing calls in the UK, incoming isn't so much of a worry as I think I'm going to accept them over ISDN. Cheers! Steve -- Steve Foy steve@narnian.org

Questions: Does anyone have a really STABLE asterisk system running about one year without need to restart the service or the SERVER ? Does anyone have a production Call Centre saled that don't lockup and is stable for 6 months ? I'm asking this questions because we have choose Asterisk for our call centre solution but, since the bugtracker only grows and people still want to stuck more

Dear All, I am trying to make the phones always talk to each other (peer to peer) using SER as a sip proxy, and incase the call is not answered we will use the voicemail of asterisk and other feautures, I have done that already, but in order to do so I found that I have to make the users dial different exten numbers, here is an example: user with exten 666 wants to call 999 . 666 dials 1999 and

back to another ? Anyway, chan_dahdi.conf : [channels] language=fr context=isdntrunk switchtype=euroisdn pridialplan=unknown prilocaldialplan=unknown internationalprefix=00 nationalprefix=0 usecallerid=yes callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0

Hi, I've set up an Asterisk as voip gatway: VOIP <-> Asterisk <-> hfc-s card <-> NTBA <-> Siemens Gigaset Dect ISDN pbx. Outgoing calls from dect handset to the world are working. Incoming calls don't even ring the handset. I'm using the dahdi driver with the zaphfc kernel module. The hfc-s card is in nt mode. The msn is set at the dect phone/base station

And I'll resend this one too. Silly scalix. --Rob -----Original Message----- From: Rob Thomas Sent: Thursday, 22 June 2006 12:19 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] zapata.conf: recent changes? Looks like you've stopped compiling libpri. All those options that are being ignored, are being ignored because they're

Dear All, I was trying to enable call forwarding, following the steps of the link on voip.org regarding this issue it doesn't work and the phone I am trying to implement on is still ringing. below is my conf in extensions.conf and the CLI output during the process. My configuration is : exten => _*5X.,1,DBput(CF/${CALLERIDNUM}=${EXTEN:2}) exten => _*5X.,2,Hangup exten =>

I love TrixBox, with the custom config files you can tweak pretty much with TrixBox too, I have at least done some. Plan to do a plain Asterisk install later, but for now I learn a lot about the config files just with TrixBox. Some things might be a bit harder with TrixBox due to some of the premade dial plans, but can get it to work :-) _____ Fra: asterisk-users-bounces@lists.digium.com

I set up the Asterisk for my company which is a business center, I will assign a specific telephone number to my client that uses my serivces. All of their incoming calls will be first picked up by the receiptionist, can I disply the company name instead of the called number on my receptionist's telephone display, so that she can answer the call with the right identity at once... Regards, ML

Hello, I have a system with two cards: a HFC-PCI ISDN and a TDM21B (2 FXO and 1 FXS), running Asterisk 1.2.4-BRIstuffed-0.3.0-PRE-1l with freePBX beta5 dialplan. I have connected an analog phone to TDM FXS port, but when I pickup the phone to make a call, Asterisk "hangs up" the call. Let me explain: In another system, when I pickup the phone, Asterisk give me tone to dial: >---

I have Asterisk 1.4.24 en a Digium TDM410 with EC and with 3 FXO modules. When I plug one PSTN-line into a FXO-port I am able to receive calls on this line and I can also make calls from an internal SIP-phone to the external PSTN-network. Still I am bothered about something that appears on the CLI when I do a reload chan_dahdi.so : asterisk*CLI> reload chan_dahdi.so -- Reloading module

Does anyone know, how I can increase the timout value for the ISDN implses, so asterisk waits for the extension on the ISDN channel? Regards, martin My zapata.conf: [trunkgroups] [channels] language=de pridialplan=local prilocaldialplan=local nationalprefix = 00 internationalprefix = 000 ; trust user provided callerid (clip no screening)? pritrustusercid = yes ; hidecallerid=no

Hi, I've a Fritz card which was working fine, recently I changed hardware and my nightmare started. Now when I call someone through the chan_capi (0.3.5 or 0.4.0) it works fine but when I receive calls I always get hungup. Can someone please give some help? Here are the logs: *CLI> -- CONNECT_IND ID=001 #0x0000 LEN=0049 Controller/PLCI/NCCI = 0x101 CIPValue

Hello, I'm having problems making international calls from our asterisk using an ISDN30 in the netherlands. Below is the zapata.conf that works for all national calls, but international calls all fail a RC=41 ; zapata.conf [trunkgroups] [channels] language=nl signalling=pri_cpe switchtype=euroisdn callerid=asreceived ;pridialplan=unknown ;prilocaldialplan=unknown immediate=no

I'm having a bit of a problem here: I have a * box with a fritz isdn card (running capi 2.0 and chan_capi) and a x100p card for testing purposes. As a proof of concept, I wanted to be able to dial into the * using the isdn line, listen to a message, and enter a 3 digit extension number. If this happens, I wanted the * box to dial out using the x100p card, into our PBX (Nortel Meridian). If

Hi, I'm using a Junghanns Quadbri ISDN card on some lines from the Austrian Telekom. Things are working, the only missing stuff is to add a "0" as a prefix to each incoming call, to make it possible to answer missed call lists. I'm using the 0 as the prefix for outside lines. I've experimented a little with the prefix settings in zapata.conf, but without success:

I've got some weird behavior on my HFC-s cards. asterisk CVS-06/26/04-21:28:35, bristuff 0.02, libpri 20040510, zaptel 20040623 When i pick up my ISDN phone on Zap5-1 ("3987") and call the external number "1901" it will do so, connect me and everything is fine. In the second, where it tries to attempt the native bridge, the audio will disappear. Using another card

Hi all, I really hope that you guys can help, because I've been tearing my hair out for the past 5 hours on this one. I have a Zaphfc (BRI) card in TE mode connected to the S-Bus of a Nortel Meridian phone system. Phone calls from the Nortel to say MSN 510 are correctly being sent to the right SIP phone. When asterisk dials say Zap/g2/224 (a Nortel internal extension) the call goes