similar to: new asterisk installation report and request for mixed voice data apps

[this message is a resend, I apologise for duplicates, but I think the original mail did not make it.] 12/3/2004 hi, I followed the instructions faithfully in http://www.voip-info.org/wiki-Asterisk+Fedora+Core+3 and the directions were very understandable and almost painless. I did miss the documentation, as "make progdocs" would not work since the package "doxygen" was not

hi, we have just received our first shipment of digium cards, FXO + FXS combinations, and collected all the hardware for our custom clone server which will house our test-bed for asterisk. I'm based in Dhaka, Bangladesh so you will understand we may not always be able to get all the hardware we need, when we need them and in the manner that other people are used to.. from a catalog or from a

Hello, I have found a "bug", I think in the way TDM400P cards handle FXO interface disconnect/re-connect problems. Normally I do keep all the wires connected from my CO / PABX quite securely, but I had a need to re-route the cable from one side of the desk to another, and I simply disconnected the RJ-45 connector and plugged it back in. THIS PROMPTLY RESULTED IN VERY VERY SCRATCHY AUDIO

hello, using a legacy PBX to access a Asterisk Zap channel (Legacy PBX FXS --> FXO application Asterisk/TDM400P) I want to be able to "flash" the asterisk pbx. However by pressing the FLASH button on the extension connected to the Legacy PBX gets me the flash features on the Legacy PBX, not on the Asterisk PBX side. I thought of using the following codes (listed below) from

Hello, I would like to parse inbound Asterisk IAX2 7-digit numbers in the form of 123-4567 and strip out the first four digits, and then dial whatever number digits remain. If I only have three digits (000-999) and have a mix of channels (ZAP, SIP, IAX2) could someone please point out how I can use a single DIAL command to just dial the extension regardless of the type of channel. .. For each

Hello, I want to use Asterisk PBX in front of my old, legacy PBX. The legacy PBX can be outfitted with caller-ID and is already able to handle Calling Party Control Signal Detection (this is a Panasonic KX-TD1232 Super Hybrid PBX. My question is how would one enable Asterisk to control the TDM400P/FXS port to provide to the /FXO CO port on the legacy PBX, support for proper answer supervision/CPC

Hi, I'm trying to make a dialup internet connection through my asterisk PBX. When I bipass the Asterisk box, I can get 51600bps. When I run through the asterisk box, I'm limited to about 21600bps. I have a TDM31B card. Any help on speeding these connections up would be good - I was on the understanding that if you bridged the channels, then the call should essentially flow straight

Hello, I thought that my Digium TDM400P would be the right hardware to support the zaptel timer, and put the following IAX.CONF entry to test, (trunk=yes) in the example below [VHAX] type=peer auth=md5 username=whoknows jitterbuffer=yes ;trunk=yes secret=terriblesecret host=4.5.6.7 qualify=1200 disallow=all allow=ulaw allow=gsm ;allow=g711u ;allow=g711a But, it didn't work. So I had to

I greatly appreciate the help at last Monday's installfest and especially am grateful to Chris and Lenny. Now I am reconfiguring the installfest computer to add Zaptel using the instructions at http://www.voip-info.org/wiki/view/Asterisk+Zaptel+Installation There is now a Digium 4 port card installed. In regards to Zaptel there are some questions on zconfig.h. Should I: #define

Anyone tried to build * + h323 to rhel3... I have to problems in the process... a) Zaptel would not build - a whole bunch of errors about kernel... b) make progdocs failed with reference to dot - check your installation. Do I need the zaptel ?? I will not be using any interface cards.. I'd like to make progdocs - any suggestions there? Latest cvs on everything.. Thanks

Not even sure how important this is considering the state of many of the online docs... I have doxygen installed as is noted for the requirements for 'make progdocs', but the make doesn't find dot. I have no idea where dot went, is or should have been... I am installing und Suse 9.0 and it's rough. If you forget something duringthe initial install, adding the package later

[resend in plain text format] > On Tue, 2004-12-07 at 09:34 -0600, asterisk-users- > > request@lists.digium.com wrote: > > > I've been struggling with a test * install for a couple months now in a > > > small office and am just about ready to give up on it. It's not that the > > > system itself is a problem. I've got everything (attendant,

> On Tue, 2004-12-07 at 09:34 -0600, asterisk-users- > > request@lists.digium.com wrote: > > > I've been struggling with a test * install for a couple months now in a > > > small office and am just about ready to give up on it. It's not that the > > > system itself is a problem. I've got everything (attendant, voicemail, > > > FXS

I'm having an issue with CDR. Basically, I expect to have all "legs" of a call having the same linkedid and differing only by the sequence value. That does happen, but I'm getting null dst values after doing an attended transfer. I'm not sure if this is a bug or I'm doing something wrong. I'm running Asterisk 13.2.0. Here's the console log, step by step: First,

Hi All, Can Asterisk dial extension which resides in the PABX? (eg. 2000) Sip Phone <-----> Asterisk <------> ATA (FXS) <------> (CO side) PABX <-----> Extension (eg. 1000) (2100 & 2101) can my sip phone call to pabx extension 1000? What will be my dial plan? I know I can connect to 1000 by

Hello all, After installing Asterisk, i have installed the docs by "make progdocs". But i don't know where to locate this documentation. please Help. Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070810/ceb95948/attachment.htm

Hi All, I would like to explain the layout that i am trying to achive. I am so helpless on this regard. So here is the story ........ " This is with regard to the setup which you can find at the "Asterisk The Future of Telephony" , chapter 11, page # 196-197, I am attaching the picture for your information. Now I am taking a challenging step to of integrate IP PBX with our

Vinicius Fontes wrote: > Hey guys, I don't know if this is the right place to ask this. I was > thinking about reporting a bug, but maybe it's better to sort out if > this is really a bug or just me being lame. > > I want to record *every* call in my Asterisk box, so I use the > MixMonitor() application like this is my extensions.conf: > > exten =>

Hi, I've updated my Asterisk 3 times with : cvs checkout -r v1-0 zaptel asterisk asterisk-addons and then do cd asterisk make clean && make && make install make samples make progdocs and then when I run Asterisk I get : Asterisk CVS-v1-0-02/11/05-01:46:25, Copyright (C) 1999-2004 Digium. Is this a bug in CVS handling or am I doing something wrong ? How to check which

All, I'm haveing a bit of trouble getting my head around H.323 and call routing with Gatekeepers, Zones and intra-zone calls - hopefully someone who is more informed in things H.323 will be able to point me in the right direction...? I already have a mature network of Asterisk boxes dotted around the UK and overseas with hundreds of extensions and our own number-plan/dial-plan in the form