similar to: Asterisk Dial Command Timeout not Accurate (not even close)

The Asterisk development team is pleased to announce new releases of our primary projects: Asterisk 1.2.8, Zaptel 1.2.6 and libpri 1.2.3. All of these releases incorporate a number of bug fixes, with the Asterisk release containing an especially large number since the last release, including some important fixes in the IAX2 channel driver. All users are encouraged to update as soon as they can to

Hello, I have a problem with the Bristuffed version of Asterisk. I have tried Bristuff-0.3.0-pre-1m,n,o,p (Asterisk 1.2.6 to 1.2.7.1) but they all have the same problem it seems: The setup: A machine with a single hfc-s PCI BRI adapter running Gentoo 2.6.15. Asterisk 1.2.0 (BRIstuffed-0.3.0-PRE-1) installed and working perfectly. Grandstream gxp-2000 as a SIP phone, and a normal mobile

Just installed fc5, installed correct kernel source, and trying to compile zaptel-1.2. Changed the link in /lib/modules/2.6.15-1.2054_FC5 to point to /usr/src/redhat/SOURCES. Like: lrwxrwxrwx 1 root root 23 May 12 15:21 build -> /usr/src/redhat/SOURCES A 'make install' still complains with: make -C /lib/modules/2.6.15-1.2054_FC5/build SUBDIRS=/usr/src/zaptel-1.2 modules make[1]:

Hi, I' ve just connected a carte X100M to my asterisk server running zaptel-1.2.5, libpri-1.2.2 and asterisk-1.2.6 on SUSE 10.0. When I make modprobe wcfxo and modprobe zaptel I haven't any error, I have also chan_zap.so module existing in /usr/lib/asterisk/modules. But, when i run ztcfg, it shows me this: Zaptel Configuration ====================== Channel map: 0 channels configured.

I managed to sort out my earlier query regarding flash times (changed delay in zapata.conf) Now, I am getting a timeout after the Flash command in an outgoing call-file based call: -- Attempting call on Zap/1/108 for 567112@demo:4 (Retry 1) > Channel Zap/1-1 was answered. -- Executing Festival("Zap/1-1", "Dialling now") in new stack == Parsing

Hi, we are still trying to properly connect a Tenovis PBX to an Asterisk server (asterisk 1.2.6, libpri 1.2.2, zaptel 1.2.5, Digium Wildcard TE110P), this time with QSIG. Calling from a Tenovis phone to a SIP phone (i.e. traditional phone -> Tenovis PBX -> QSIG -> Asterisk -> SIP phone) works with the following messages: --- Don't know what to do if second ROSE component is of

Hi, does anyone know about a plugin that allows dialling a contact from the phonebook of evolution or T-bird? -- Alexander Topolanek http://www.topolanek.at

Hi list! I have some sip phones and Sipura ATA 2000's. However after dialling a number I need to dial a # to control a device. When I dial # Asterisk kicks in and puts the call on hold. How can I change this? Thx!! Remco

Dear all, How may I configure my extensions.conf so that only the boss's secretary can call the boss through his extension, all others when dial his extension only makes the boss's secretary phone ring, not his. If she wants, she can transfer the incoming call to the boss dialling his extension. I've tried the following, but it doesn't work: exten =>

Hi all, I hope I'm not starting a flame war. I'm installing Centos 4.2 for a mail server. I've been experimenting with, Centos before, and now wanting to put it into a real action as mail server. Can someone pls tell me the up and down between Centos and FC? I'm familiar with FC4 and some things that I like from it: 1. The available packages are abundant. Very easy to find

Good day, Hi! i've finish up setting * for my company and they are working reallly great, but i notice when i try to call to mobile phone, i can see the zap channels is bridging successfully but i hear nothing except for a long dialtone like tone, but calling to a regular pots line is working perfectly, could this be related to telco issues? or some tweaks to zapata.conf p.s. but i'm not

Is it possible to make Asterisk jump into action and play a sound file as soon as a handset is lifted, instead of providing a dialling tone and waiting for the user to dial an extension? -- AJS Answers come *after* questions.

Hello Maybe a silly question, but after some searching couldn't find answer. Is there a number I can dial to pickup and listen to my voicemail messages on my SIP phone? I am used to eg dialling *17 to pickup my voicemail messages on Avaya system? Angus -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi All, I am using Linux 2.6.15 kernel. I want to create 4094 bridge interface but I am able to see only 32 bridge interfaces with the help of "brctl show". I am able see other interfaces into /proc/net/dev but brctl is not showing more then 32. It is also not allowing any operations (add/del) over those interfaces. Is it correct behaviour or I have do some changes in some settings.?

Hi, We have been migrating our PBX system from Asterisk 1.8 and chan_sip to Asterisk 13 and chan_pjsip. Things are mostly, ok, but now I have stumbled on a behaviour difference I don't like. With chan_pjsip when a phone went unexpectedly offline (Ethernet cable disconnected) Asterisk would detect this quickly (through the 'qualify' pings), mark the phone as 'Unavailable' and

I'm not sure if I have a configuration problem or not. I am unable to dial out. When I try to dial in I can hear the phone ring on the dialling phone but Asterisk does not register anything. In zaptel.conf I have loadzone = au defaultzone=au fxsks=1 In zapata.conf language=au context=from-pstn When I do: zap show channels I get: Chan Extension Context Language

Hi, I have an asterisk installation with 2 E1 cards Software version is Asterisk 1.2.6 Libpri 1.2.2 Zaptel 1.2.5 I'm having problem with fax transmission, let me explain better my setup: My fist TE110P E1 card is connected to the telco line the second TE110P E1 one to an Nexspan PBX so the server is basically sitting between the line, and the pbx. every call coming from the line is

I am instaling asterisk on Fedora core 3. I have instaled zaptel-1.2.3, libpri-1.2.2, but when I am instaling (make install) asterisk I have the following error: .................... _GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -fomit-frame-pointer -fPIC -c -o app_zapscan.o app_zapscan.c gcc -shared -Xlinker -x -o app_zapscan.so app_zapscan.o gcc -pipe -Wall

Hi, Do the callers in USA dialling from USA Telco lines always have to prefix the CITY/AREA code with "1" in order To successfully make a call to other USA destinations? ---- I have not been to USA (yet) :) Ta SJ

Dear All, I was wondering will there be any problems or changes that I will need to do to compile the current Asterisk(1.2.7)/Zaptel(1.2.5)/Libpri(1.2.2) source from www.asterisk.org into a 64-bit binaries? I am currently using the following hardware for my new server. CPU: Pentium D 930 3.0 GHz Mobo: Intel D945PSN Motherboard RAM: 512MB 533MHz DDR-2 Drive: SATA II Seagate 160GB Card: TE406