similar to: executing an agi script after a successful Dial

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,

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

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

Hi All, First of all, please no flames about the footer of this message. It's being inserted into all outgoing email by the server here without asking me personally first once I send out email. Since I've been very interested in setting up an Asterisk PBX + IVR + Voicemail system to support a small business and a foundation for quite some time now, in order to replace their current ISDN

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

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

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

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, 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)

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

Hello all, I'm trying to get CAPI to work with two B channels (AVM B1 PCMCIA) on a P4 2GHz (linux kernel 2.4.21) system. All are ok with just one B channel. But when I open a second B chan, the sound is choppy, with too long gaps, and the CPU load is too high (~50%). On the Asterisk's console I get these messages: -- Executing Dial("H323:4478",

Having ISDN FXOs and a bunch of ISDN DECT base stations, I went looking for a CAPI compatible active BRI adapter capable of NT mode. I have found that the 'Eicon Diva Server 4BRI' fulfills theses requirements. However, at EUR 1387 for four BRI interfaces, it does not come cheap. Are there other users of similar hardware that can point me to alternatives ?

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,

After apt-get update && apt-get upgrade -y wget http://www.junghanns.net/asterisk/downloads/chan_capi.0.2.5c.tar.gz tar xfvz chan_capi.02.5c.tar.gz cd chan_capi-0.2.5c make && make install shutdown -r now asterisk seg faults upon calling in via ISDN. Any ideas are greatly appreciated. rgds pos

Hello, i have asterisk with 2 internal isdn cards - handled by isdn4linux and i need to setup whol system like this route some call beggins with 0 or 00 - long distance through first card, route calls to mobile network via second card ( tehere is isdn gsm gateway connected).how i can do this using only isdn4linux (/dev/ttyi) ? regards Marian -- SUNTEQ s. r. o. Bojnicka cesta 35 # Prievidza

hi all I'm trying to setup a dlink dph-100h phone (actually a dph-100m but with the h.323 software) with asterisk and chan_h323. AFACS, the dph-100h software can only be configured to point to a gatekeeper. I know I don't need to do this, but it's a test before I setup my Symbol Netvision phones (I don't have an access point for them now). First, this is what I have, and what

Hi. I just tried updating asterisk and I guess I broke something. Here's the log: Unable to find a path from G729A to SLINR Unable to find a path from ULAW to G729A Any ideas on what I should try? I tried nuking all the zaptel stuff in the system and the source and started over agian. Also nuked the asterisk config files.... I saw this asked once before but there was no reply :-/