similar to: Is there a problem when we want to transfer an incoming call to an external phone number

What do you want to knwo exactly ? The way I use to transfer the call is the same if it was an internal extension, I use the Dial command using the trunk to go outside ... Ask me the info you need ... Thanks David -----Message d'origine----- De : Bob Goddard [mailto:asterisk@bgcomp.co.uk] Envoy? : mardi 14 juin 2005 12:52 ? : Asterisk Users Mailing List - Non-Commercial Discussion Objet :

I have the same problem in a box with 2 HFC-PCI, but i already remove the row in modprobe.conf and load the module manually. Both cards works fine Any idea ? Giordano -----Messaggio originale----- Da: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] Per conto di Julian J. M. Inviato: luned? 27 giugno 2005 12.04 A:

Hi, I want to use the cdr to record the call log to my Microsoft SQL Server using unixodbc and freetds.... but when I compile, I've got this message.... Does anyone have the same problem and/or know how to solve it ? Thanks Baste regards David Masure make[1]: Entering directory `/usr/src/asterisk/bristuff-0.2.0-RC7k/asterisk-1.0.6/cdr' gcc -pipe -Wall -Wstrict-prototypes

Hi, Bristuff works great with HFC card... your compilation problem may come from your kernel configuration... You should check this doc, at least, for the redhat config : http://www.automated.it/guidetoasterisk.htm Then, installation of Bristuff works like as charm ! Bye David Masure -----Message d'origine----- De : vdasilva [mailto:aquamoon@wol.co.za] Envoy? : mardi 28 juin 2005 09:01

Hi! I have just installed Redhat 9 and Asterisk to my computer, and now i have problems with my non-zaptel Card, I don't know how to set it up since all instructions are for digium's hardware. I have searched from the Internet for hours now, can you help me to understand all this HFC-s thing and how it is related to CAPI, ISDN4Linux, bristuff and so on. I have to say that I am not so

Hi Folks, I've been experiencing something very strange... When I want to listen to call between a SIP phone and a Zap Channel, I can listen a with a nice audio quality. When it comes to record using the monitor command, I just have a wav file which is completely noisy and I can't hear the conversation. Questions ? 1) Is there a way to specifiy the audio quality using monitor ?

Hi, Question... Is there a way to receive two phone calls on the same phone, or, for example to receive a phone call, put the call in stand-by and then make another call and finally, why not put them all together in conference... Thanks David Masure -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all, I have an asterisk box set up with a bri card (using zaphfc). I have a bri cellular gateway connected to it beacuse I'd like to route all my cellular calls through that gateway. The probel I encounter is that when trying to dial a phone number, I've the message : unable to create a zap channel. My card is normally well configured because when connected to the NT, It works

Hello, I've added a ZAPHFC card to my CAPI based system. Calls coming in via ZAPHFC do not forward the caller id to the SIP phones. Calls coming in via CAPI do forward the caller id to the SIP phones. Any and all help is greatly appreciated. The (hopefully relevant) conf file excerpts are: extensions.conf =============== exten => 807440,1,Answer exten => 807440,2,Noop exten =>

Same problem here: if call come over ISDN PRI and it is for a SIP phone that equals to strong echo situation, at the SIP end. Interestingly this doesn't happen on all calls but it does on 95% of them. Asterisk load at that moment is insignificant - 1 to 2 calls. I have tried with all possible echo cancellers in zconfig.h, with and without MMX, and with and without CFLAGS+=-march=i686 in

I have ISDN BRI line (Point-to-multipoint), with HFC card and bristuff/zaphfc driver. I also have TA attached to NT, with analog phones and modem on it. When I'm dialling through Asterisk/bristuff, and in the same time TA have some conversation (or maybe modem link) on channel 1, I can hear that conversation (or modem) very short period (0.2sec), and I also disturbing that conversation

Hi folks ! I bought two sipura 841 phones. I used to have GN Netcom headset which I connect instead of the handset. The problem is that I don't have any sound coming out the headset and I can't speak neither ! I'am located in France and I was wondering if the cabling in the sipura and in the headset is the same (I mean the order of the cables) or maybe is there something else to

Does anyone test Asterisk 1.2.X + bristuff-0.3.X and TDM card? We can't get it to work. -David -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Julian J. M. Sent: Friday, March 31, 2006 1:44 AM To: Chris Earle; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Re: BRI

Hi All, It's broken !! (drat) Asterisk if failing to load with the following error (taken from end of /var/log/asterisk/full) after adding bristuff. Can anyone help please? Jul 17 19:57:54 VERBOSE[2503]: == Registered channel type 'Phone' (Standard Linux Telephony API Driver) Jul 17 19:57:54 VERBOSE[2503]: [chan_zap.so]Jul 17 19:57:54 VERBOSE[2503]: [chan_zap.so] =>

Hi!I've installed Asterisk@home and I have a ISDN card,(Cologne Chip Design GmbH ISDN network controller [HFC-PCI](rev 0.2) This is how I installed bristuff: how to install hfc card after unload asterisk and amportal whit amportal stop type "setup" unselect zaptel in system service... and set the lan --->reboot<--- cd /usr/src wget

Have asterisk@home 1.2.1 The server is on an internal network eg 10.10.10.10 It is NAT'd 1:1 via Checkpoint firewall to external public IP eg 50.50.50.50 The remote IAX2 phone (ATCOM320) is configured to call 50.50.50.50 on extension 1055. Outbound calls to 1055 work perfectly. Inbound calls from 1055 get picked up as if it were an external call (see below) and goes straight to the ring

Hello I have asterisk running in Red Hat 9 with a cheap HFC card on i4l. I have choppy sound problems sometimes, and echo problems often. I am using a 2 port Grandstream ATA, Grandstream BT and a Grandstream GPX-2000 I read that changing to BriStuff will fix the echo problems, but have also read other users say that the only way they solved the echo/choppy sound problems was using a Fritz ISDN

Hello, I have about 10 Asterisk PBX in production with Bristuff-0.2.0-RC8q (asterisk 1.0.10) and I want to use Bristuff-0.3 now for the new PBX I am going to set up. With Bristuff-0.2.0-RC8q the ISDN lines are working fine, but the new version of Asterisk add some nice features. All these PBX are in France with France Telecom lines. When I use the new version after about an hour with

This is my first posting to the list. Thus hopefully the question has not been asked several times before... I'm trying to install asterisk on a PC using Debian Sarge but without any real success... Problem: I may get the Asterisk packet (Debian version) installed using 'apt-get install asterisk', but that does not include the bristuff from Junghanns

Again trouble compiling bristuff-0.3.0-PRE-1q with the florz patch on a x86_64 box (I guess nobody is using x86_64 platform or is able to fix this themselves?) First of all when bristuff is downloaded and compile is started it appears that the bristuff Makefiles are badly broken. The asterisk Makefiles all do see to find the kernel sources on a RHEL4 box in the proper directory, the pure