similar to: zaphfc and T0 ISDN to Alcatel PBX

Hi, I solved the Problem: the context defined in zaptel.conf: context = zapincoming was not in extensions.conf, so nothing worked. ( I assumend this context is only necessary for calls from zaptel, not to it) This may be a bug: in the verbose log I found nothing like "[chan_zap] Could not initialize ..., due to ....", this had helped alot and had saved alot of hours (-;

Hi! I'm struggeling had to get a hfc pci card working in NT-Mode. I've installed a very basic test system with just one hfc pci card connected to an ISDN-phone over a modified NTBA as described on jollies homepage. I commpiled asterisk as descibed in kape's INSTALL file that comes with 'bri_stuff'. I have not modified the config fieles yet. After connecting to asterisk by

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

Please forgive me, if I misunderstand the problem completely. Following instructions in several german blogs, I want to configure Asterisk with a hfc-pci card, an old NTBA and an ISDN phone as a SIP device. It seems that I have to set signalling in zapata.conf to bri_net_ptmp. When I do this, Asterisk will hang if started with -vvvvc, the last lines of output being: [res_features.so] =>

Hello, I have asterisk@home with zaphfc patch applied (http://dondisperato.blogspot.com/), but I can not make call to legacy PBX (Alcatel 4400). I can only accept incoming calls. I am dialing with this: exten => 202,1,Dial(Zap/g1/242) --- asterisk1*CLI> bri debug span 1 Enabled debugging on span 1 -- Executing Dial("SIP/201-4678", "Zap/g1/242") in new stack

Hello all, if I try to call from one phone on the internal S0 to another on the same S0 using zaphfc, the bus is hung up. The called phone is ringing, but I can't talk from one phone to the other. The error I get is: -- Executing Dial("Zap/2-1", "ZAP/1/55|15|tr") in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called 1/55 -- Channel 0/1,

Hi, I've been learning asterisk for a couple of weeks now - and it worked for me as faar as standard configurations where concerned (sip/iax outbound/isdn4linux & capi with AVM Fritz!, Digium X100P FXO). Now I recently I'evaluating to use asterisk as a replacemnt for our companies (15 employees) legacy pbx system and I'm experiencing multiple problems with the hfc isdn cards:

Hi. I'm running some asterisk boxes on different sites, some equipped with a couple of ZapHFC cards, others with Junghanns quadBRI cards. All boxes were compiled with Asterisk 1.2.10 (libpri 1.2.3 zaptel 1.2.6) and bristuff 0.3.0 pre 1s, distribution is Fedora Core 4 with kernel 2.6.17.3 The cards are connected to Telecom Italia's NT1/NT1+ S/T lines; some of them are point-to-point,

Hi guys, I need to interconnect an asterisk server (1.4.17) with an alcatel pbx, because i want that an asterisk extension can call an alcatel extension and viceversa. I need to do this throug a NT port (of the digium card) for technical reasons (linked with alcatel hardware). Card installation was fine, but i'm not sure of the configuration i choose. In /etc/misdn-init.conf i put my

Hi, Does anyone have any experience connecting asterisk to alcatel 4200 series pbx with bri cards? Does it should work with asterisk bri in NT mode, and alcatel bri with TE mode? Cheers, Igor Neves.

Hello list. I'm going te explain my trouble. I have my asterisk with a TDM400P with 4 FXS channels. Two ports are connected to a Panasonic PBX (it's working fine), and others two ports are connected to an Alcatel 4200 PBX (but it doesn't anwer). I connected to a CO port (where i had a pstn line). When I call to the Alcatel PBX, the asterisk show me in it console that es ringing but

I've this error How i can find the problem? Apr 23 12:24:43 WARNING[131081]: PRI: received TEI check request for TEI = 89 Apr 23 12:24:47 WARNING[131081]: PRI: received TEI check request for TEI = 89 Apr 23 12:24:48 WARNING[131081]: PRI: !! Got a UA, but i'm in state 1 Apr 23 12:24:53 WARNING[131081]: PRI: received TEI check request for TEI = 89 Apr 23 12:25:02 WARNING[131081]: PRI:

Hello, I have 2 analogue ports on an Alcatel PBX patched to 2 FXO ports on my <mailto:*@home> *@home 2.8 running on top of CentOS. Both FXO Ports are on ONE Digium card. What I would like is: If someone calls extn 281 on my Alcatel PBX it is routed through to Extn 233 on my * thruogh FXO port/module 4 If someone calls extn 282 on my Alcatel PBX it is routed through to Extn 234 on my *

Hello everyone, I try to connect the Asterisk PBX Server on my NTBA (NT1) - actually its "hard"-connected on the box, but I've tried a crosslink-ISDN cable, too. My ISDN-Phone is connected on one port (of two) S0-Ports on NTBA. When I start Asterisk and want to make a call I'll geht a lot of this messages --> 040827-034639 WARNING[213005]: chan_zap.c:6902 zt_pri_error:

Hi there I have a TE110P card fitted in my linux box running : Linux version 2.6.9-5.ELsmp (bhcompile@decompose.build.redhat.com) (gcc version 3.4.3 20041212 (Red Hat 3.4.3-9.EL4)) #1 SMP Wed Jan 5 19:30:39 EST 2005 I followed the installation steps on digium website...no errors reported. The modules seem to have loaded...here's what lsmod shows: Module Size Used by

Hello everybody, I just buy a X101p clone and i'm new in asterisk. Here is my configuration : ISDN line ---- Alcatel ----PSTN ext 68-----Asterisk with X101p clone ------sip phone ext 200 - 203 ||| ISDN phones ext 60-67 >From sip phone to ext 60-67 it works. 9+extnumber >From sip phone to Land lines it works. 9+0+phone number >From ext

Hi, I have a partial working installation with zaphfc. Incoming call : For incoming call, seems work fine. But the sound is very bad with bounce short crashing sound. Same sound with echo cancel off or on. SDA work fine. Another problem, it's seems that's zaphfc don't reset correctly the line. I have one of my D channel how was busy even after stop communication. Outgoing call :

Hi Running slackware 9.1 with compiled kernel patched to 2.6.6 running ok. Having problems compiling zaphfc.c First thing was the can't find "irq_vectors.h", but "solved" by changing the #include "irq_vectors.h" to #include "mach-default/irq_vectors.h" in /usr/src/linux/include/asm/ directory. But I still have errors with "structure has no

Hello, AFAIK the feature CD (call deflection) is only possible on point-to-multipoint links, is this correct? I've heard about the feature "partial rerouting" which should do the same on point-to-point-links. Is this implemented in either bristuff or chan-capi(-cm)? Thanks in advance, Karsten

-----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Zoltan Szecsei Sent: zaterdag 16 juli 2005 12:54 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Multiple ISDN BRI Units with Asterisk usingBristuff zaphfc in NT mode? >> Maybe this is rather a hardware question,