similar to: Zapata Telephony interface and torisa module error

Hi. We have people with two or more sip phones. One wireless and one wired. So this is the case: Person A with two phones wants to have a queue for his incoming calls. So when he answers one of the two phones, the other phone should not ring. But when he isn't talking in any of the phones, they both should ring. Does that make any sense? This what I have for people with only one

Hi All, I am a newbie to Asterisk. Presently I am working with Asterisk 1.4.17 and using it to make SIP calls. I have a configuration of Asterisk which serves the users in a particular domain, say internal.com I would like to make a SIP call from bob at internal.com to charles at external.com I have added the following lines in extensions.conf exten =>

hi all i've just setup an * box and want to test voip calling, initially from sip user to sip user... local sip users can call each other, no issues. problem arises when i try and call a remote sip account, my * box always returns "SIP/2.0 404 Not Found" any ideas ?

What is the variable like $peerip to get the registered ip address for a peer Regards ********************************************* No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do

Hey all, after hooking up some Plantronics to some Snom's (3 320's 1 360), I noticed my client is having some form of feed back on the phone. Because of Snom's "inner oddities" this is how I got it to work. Plantronic --> RJ11 --> SnomHandset Port (on Snom Base) Handset --> Plantronic jack (bottom base in the front) If I placed Plantronic(RJ11) --> Snom's

?Hola! torisa.c (from zaptel 0.50) says: #define MASTERVAL 0x41 /* Enable E1 master clock on Rev. B board */ Does that mean that there is a torisa revB? If so, are the schematics downloadable from somewhere? (www.zapatatelephony.org only has Rev. A ones) Does the Rev. A support E1? (just changing the DS2151 for DS2153, the DS crystal and closing the E1 jumpers, or something else is

Followed the "instructions" on voip-info.org regardinging fedora FC2, making Zaptel seems to work fine, however when I modprobe I get this. It looks like a version mismatch somehow. Ideas? If this ooc, sorry first post here :-) modprobe tor2 WARNING: Error inserting zaptel (/lib/modules/2.6.5-1.358smp/misc/zaptel.ko): Invalid module format WARNING: Error inserting zaptel

?Hola! I'm trying to understand the torisa design and i'm mistified by the upper part of the third page (generation of ALE and DWR from MEMWR and CS2..5) Can somebody explain me how the magic there works? Why can't just do ALE = ~(CS3 | CS5) and DWR = MEMWR ? (DS app note 326/452, "DS2141A, DS2143, DS2151, DS2153 Interfacing to a Nonmultiplexed Bus", at

/bin/sh: -c: line 0: syntax error near unexpected token `;' /bin/sh: -c: line 0: `if [ -n "" ]; then if [ -f ]; then mv -f .bak ; fi; cat .bak | grep -v "alias char-major-250" | grep -v "post-install torisa /sbin/ztcfg" | grep -v "post-install wcfxsusb /sbin/ztcfg" | grep -v "alias wctdm" | grep -v "post-install wctdm

Good Day list, My head is pounding from google overload. Does anyone know if there is any way to park a call and specify the ("context", "Extension", and Priority) of where to call back to if the call times out and is parked too long? ParkAndAnnounce does this feature, however I do not like that it has to call you back to announce the location. I want my cake and eat it

Hi, I've compiled Asterisk cvs stable (few days ago) unde Suse 9.2 without any problems. We're using te110p and wcte11xp module that is autoloaded by Suse 9.2. Card goes green after reboot, but this meesages appear in logs: Mar 22 11:28:51 linux kernel: Zapata Telephony Interface Registered on major 196 Mar 22 11:28:51 linux modprobe: FATAL: Error inserting torisa

Hello all After install the Asterisk, setup the configuration file in the sip.conf and extensions.conf and type "asterisk" with start program. I don't have line card yet. But I found following error in the /var/log/messages. localhost insmod: /lib/modules/2.4.18-5/misc/torisa.o: init_module: Input/output error localhost insmod: /lib/modules/2.4.18-5/misc/torisa.o: insmod

Hi, Asterisk Version : 1.2.15 Card : TDM11B (1 x FXO , 1 x FXS) I have internal dialling working okay SIP->ZAP (analogue phone) and ZAP (analogue phone) -> SIP. The problem comes when I try and make a outbound call. Here is my extensions.conf :- Code: [incoming] exten => s,1,GoToIfTime(17:00-09:00\mon-fri\*\*?outofhours|s,1) exten => s,2,GoToIfTime(*\sat-sun\*\*?outofhours|s,1)

Hi guys, Thanks for your answers on my two questions yesterday. That's exactly what I was looking for, sorry for not noticing it myself, but I'm still getting acclimated to Asterisk and even Linux--from what I see so far, I love it. I've got another one now. Since my Asterisk install and configuration is fairly stable at this point, I'm interested it ensuring that during

On 11/09/2020 18:30, bobby wrote: > I am now running 2.3.11.3 (502c39af9), and am still getting these > messages. OK, good. What is your current version and configuration (output from `dovecot -n`) Anything interesting in the logs? Any idea which deliveries are causing this? Can you obtain an LMTP protocol log for such deliveries? Regards, Stephan. > On Fri, Sep 11, 2020 at 10:52

i have error in install module of tor2. but it compile good. what happen ? ivr2:/usr/src/zaptel # make clean; make install rm -f torisatool makefw tor2fw.h rm -f zttool rm -f *.o ztcfg tzdriver sethdlc rm -f zonedata.lo tonezone.lo libtonezone.so.1.0 *.lo rm -f gendigits tones.h rm -f libtonezone* rm -f tor2ee rm -f core cc -I. -O4 -g -Wall -DBUILDING_TONEZONE -DSTANDALONE_ZAPATA -c -o

Hi, Is it possible to pick the context of a call from chan_sip based on the domain of the To: header of the INVUTE? I've had a quick look throught he code and can't see anything, I want to use the voicemail virtual hosting with chan_sip. Can the sip domain be picked out with a global in extensions.conf? This woud also solve my problem. If not is there any specifc reason/restriction

(I'm producing a writeup on all the things one run into building an * box. Which after a couple of boxes is showing up aplenty.) So this uses the working h/w and config's from another box. All I do is build in a new computer. Loading the modules I got: __________________________________________________ Zaptel Configuration ====================== Channel map: Channel 01: FXS

Hello - any help is greatly appreciated. I am trying to compile zaptel on debian 2.4.26-1-386. I have a single X100P card installed. When I run '/usr/src/zaptel/make clean;make install' I get the following: depmod: *** Unresolved symbols in /lib/modules/2.4.26-1-386/misc/torisa.o depmod: *** Unresolved symbols in /lib/modules/2.4.26-1-386/misc/zaptel.o /sbin/depmod -a depmod: ***

Is there a way to pass variables/arguments to the h extension ? for example : [default] exten => _1098933X.,1,NoOp(CARRIER TWT->TIM, EXTEN: ${EXTEN}}, SIPCALLID: ${SIPCALLID}, SIPDOMAIN: ${SIPDOMAIN}) exten => _1098933X.,2,SetVar(_PROVA="bla") [lot of stuff, agi, goto, tricks and magic that happens] exten => _1098933X.,10,Dial(${CHAN_DEST},,L(3600000:3599900)) <-