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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, 10 Fe...

I know this has come up before but... Has anyone done anything to get an IAX client built on Windows? I thought someone had started one, but I haven't heard anything about it since - and that was months ago? Anyone have any idea what the status is? -- Ron Gage - Saginaw, Michigan I am looking for work - resume at http://www.rongage.org/resume.doc Electrical Engineering, Linux Programming,

Hi I want to set up * as a conference bridge. I would like to be able to conference is SIP calls (up to 12) I am looking through all available documentation for * to get info on how it is done. No luck so far. Can somebody direct me to the info in this subject? Thank you Serge _________________________________________________________________ Protect your PC - get McAfee.com VirusScan Online

...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 port BRI interface going GA ! Regards, Hans P.S. This is excellent ! -----Original Message----- From: Klaus-Peter Junghanns [mailto:kpj@junghanns.net] Sent: Thursday, November 06, 2003 4:35 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] ISDN PBX + IVR + Voicemail Configuration - Sanity Check ... Hi Hans, Am Don, 2003-11-06 um 15.58 schrieb Vledder, Hans: > P.S. He who comes up with clean internal ISDN bu...

...mgcp If one waits until the last one rings, then hangup, everything is fine. If one waits until the last one answers, then hangup, everything is fine, too. Any hints? mgcp debug on: -- Executing AGI("Zap/7-1", "nuller.agi") in new stack -- Launched AGI Script /home/kpj/pbx/var/lib/asterisk/agi-bin/nuller.agi -- Accepting call from '01635571857' to '8551' on channel 0/1, span 3 -- AGI Script nuller.agi completed, returning 0 -- Executing Dial("Zap/7-1", "MGCP/aaln/1@192.168.1.26||") in new stack -- MGCP mgcp_requ...

...300 bucks per day, and no insurance company would cover them >in a data center inviroment. since we was going to produce our own cards, i >thought i would share with the rest of the world our findings. >dwayne >----- Original Message ----- >From: "Klaus-Peter Junghanns" <kpj@junghanns.net> >To: <asterisk-users@lists.digium.com> >Sent: Thursday, April 03, 2003 9:58 AM >Subject: Re: [Asterisk-Users] CE certification for Europe > > > > >>Hi d hintion, >> >>hmmm...getting approvals for europe isnt that easy. >>because...

I would like to run an agi script (to calculate the cost of a long distance or international call) right after I execute a Dial app. Can this be configured in extensions.conf? It seems the entries right after a Dial app get executed only if the Dial app was executed unsucessfully. Would I have to execute the dial app from the agi script? Thanks in advance. Dan -------------- next part

Hi, If anyone is looking for a small Asterisk installation I have managed to get it down to 296MB (If you remove the kernel source code.. could probably be made smaller if some of the devel packages and asterisk source is removed as well.) To do it I used Trustix Secure Linux 2.0 (http://www.trustix.net), did a minimum install with ssh support(92MB) and then added the required packages

Hello, We have been trying to add asterisk to our Ascom Ascotel 2050 PBX. We have a AVM Fritz!PCI Card connected to an S0 bus extension from the PBX. The fritz card is configured to use chan_capi, and we can make calls SIP->SIP SIP->PBX extension PBX extension->SIP all successfully, we have assigned more than one PBX extension number to the S0 port in the Ascom PBX (it has 8 positions)

I'm currently using the SJPhone softphone with asterisk for remote SIP. When I dial into the voicemail, and attempt to pass the extension, I "hear" the sounds, but asterisk is not receiving any DTMF signals. If I use the Estera softphone, asterisk does receive the DTMF signals. Normally, I'd just say "Use the Estera" softphone to myself, but that's not an option,

hi all talking to this friend if mine just now, I realised I had no 'micky mouse' sound as eaerlier with chan_capi and AVM C2. We talked in 1 hour and 18 minutes without encountering any problems whatsoever. This was him calling me, but I don't know if that's important. roy -- Roy Sigurd Karlsbakk, Datavaktmester ProntoTV AS - http://www.pronto.tv/ Tel: +47 9801 3356

I have an old A1 ISDN card that we used to use in the mid 1990's for the delivery of data to our ISP I have seen some users have been using them, with respect to doing web searches. Can someone please let me know what drivers I require to use this card under debian and with asterix, and what the limitations are of using this card. I dont want to make calls but do inbound using it. The

hi all is it hard/possible to move the following from chan_capi_pvt.h into a setting (preferably global) in capi.conf? #define AST_CAPI_NATIONAL_PREF "0" #define AST_CAPI_INTERNAT_PREF "00" and ... Is it hard to move or copy the txgain and rxgain to [global], either as a given 'default' if nothing's set in the interfaces, or as a overall

hi all it seems like whatever I do, I can't use more than 1 BRI on my AVM C2 with chan_capi. Both channels seem to work, but not at the same time. And - yes - they're connected to different NT boxes :) Any ideas? kapejod? -- Roy Sigurd Karlsbakk, Datavaktmester ProntoTV AS - http://www.pronto.tv/ Tel: +47 9801 3356 Computers are like air conditioners. They stop working when you open

I have been using chan_capi and a Fritz AVM (passive) card for a little while now.. So far so good.. There are still a couple of things I am not totally sure about.. First.. What is the difference between "softdtmf=0" and "softdtmf=1"?? Which is the preffered/recommended option?? Second.. What is Early B3 all about?? Should I be using it in my dial string?? Which is the

hello, does anybody know how to setup avm fritz pci card in p2p mode ? regards marian -- SUNTEQ s. r. o. Hviezdoslavova 9 # Prievidza # 971 04 # Slovak republic Tel: +421-46-5430 754 # Fax: +421-46-5439 144 http://www.sunteq.sk/ ------------------------------------------------------------ A mind is like a parachute... it only works when it's open.

Hello. Although I can hear the demo etc. now, I notice during asterisk launch I get :- [chan_oss.so] => (OSS Console Channel Driver) == Console is full duplex == Registered channel type 'Console' (OSS Console Channel Driver) == Parsing '/etc/asterisk/oss.conf': Found WARNING[98311]: File chan_oss.c, Line 232 (sound_thread): Read error on sound device: Resource

...is used to redirect the extra channel regardless of the Priority and ExtraPriority properties. A patch for manager.c is available at http://www.reucon.net/~srt/bristuff_redirect.patch as a result to a bug filed against Asterisk-Java at http://jira.reucon.org/browse/AJ-34 I've sent a notice to kpj. =Stefan -- reuter network consulting Neusser Str. 110 50760 Koeln Germany Telefon: +49 221 1305699-0 Telefax: +49 221 1305699-90 E-Mail: srt@reucon.net Jabber: srt@jabber.reucon.net -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: applic...

Hi, I'm using * with SIP in our LAN and CAPI for external phone calls. Everything works fine (SIP to SIP calls are ok and external incomming calls are ok too (CAPI to SIP)) exept the CAPI to SIP communication. When I try to make an external call from a SIP client, the (ext) phone rings (while the SIP client display "trying...") but when I hangup I don't get any communication and

...e >sterisk, ie >${UUS1}, on an incoming call it would contain the message, and on an >outgoing call it could be set to the message we would like to send. > >Sorry i can't be of more help, but if you really want this feature, i >suggest you >talk to Klaus-Peter Junghanns <kpj@junghanns.net>, i think this would be >only a few lines of code. > >Regards, > >Christoph