search for: sprackett

...ited non-zero on 'SIP/mitel-fe17' THe PSTN gateway is siezing the trunk and dialing the call. The native bridging seems to be the point of failure. The caller (another sip set) gets hung up on and the called pary hears dead air. A sip debug log of the scenario is available here: http://sprackett.com/asterisk.txt extensions.conf snippet: [trunks-local] ; local 7 digit dialing exten => _NXXXXXX,1,Dial(SIP/${EXTEN}@mediatrix-1204) exten => _NXXXXXX,2,Congestion [internal] include => trunks-local sip.conf snippet: [mediatrix-1204] type=peer host=172.20.16.7 mask=255.255.255.255...

Hi This is hard work :) I have read the Asterisk Handbook, BudgeTone User Manual, Andy Powell's useful notes, Zac Sprackett's Asterisk Resource Pages and more. I am not a linux newbie but am new to Asterisk. I have failed to find any docs that explain how to get a very very simple, minimal, system up and I am trying to get the following to work: 2 BudgePhone 102D connected on a LAN to a linux RH9 server running As...

It would be nice to see a real "extensions.conf" from a live business operation, every extensions.conf I've seen posted or been able to dig up so far would fail bad in a live business operation. I just have the beginings of mine and would like to make sure I don't miss anything. Most extensions.conf files I've seen wouldn't even let you dial "911" in

Im having a trouble understanding the config files setup even with some documentation ive read such as the handbook, maybe im just illiterate. Anyway do you think some one can point me to some examples of real config files. Such as IAX, Extensions, and Sip. I just cant grasp the concept for some reason. If someone would like to help me out, maybe even explain one on one? Thanks a lot

Hi, I'm struggling to get busydetect to work. I'm using asterisk 1.2.1 and a digium TDM04B (4 port FXO) card. I've set busydetect=yes, busycount=6 and busypattern=300,200 in zapata.conf and i've modified zondata.c with a busy setting of 620+480, 300/200 which is the busysignal received from Korea Telecom. Asterisk isn't detecting the busy signal and doesn't hangup.

...alls. An agent logs in=20 > indicating that he's now available to take calls. Asterisk then sends calls= > to each agent as they are free to take a call. > > Members are those calling in and piling up in the queue, waiting for an agent > to answer. ...hm, then why http://www.sprackett.com/asterisk/conf/queues.conf says "agents or members who are listening to more than one queue can differentiated how they should engage the customer". Also the same configuration file list agents as *members* of the queue: member => Agent/01 member => Agent/02 >>...

Ok. I've done more research and testing and here are the details. It is using the dialparties.agi (http://www.sprackett.com/asterisk/) file to dial . Originally with the dial options "tr". I changed the options to "Tt" but no change. Transferring internal extensions between each other works fine. Example: 201 calls 202. 201 transfers 202 to 203. Transferring the IAX trunk to other internal...

Would anyone mind sending me a working set of config files for asterisk and their softphone settings? I am really looking for very basic setup stuff as I just want to show that the system works to my management before they will allow me to spend the money on phones and a telco card. Server is a redhat 7.3 w/ 512 RAM and dual 550's. Asterisk tar ball has been laid down, configured, make,

...eived group=1 channel=4 --------------------------- My extensions.conf is as follows: ------------------------------------------------------------------------------------------------------ ; Asterisk Management Portal (AMP) ; Copyright (C) 2004 Coalescent Systems Inc ; dialparties.agi (http://www.sprackett.com/asterisk/) ; Asterisk::AGI (http://asterisk.gnuinter.net/) ; gsm (http://www.ibiblio.org/pub/Linux/utils/compress/!INDEX.short.html) ; loligo sounds (http://www.loligo.com/asterisk/sounds/) ; mpg123 (http://voip-info.org/wiki-Asterisk+config+musiconhold.conf) ; include part generated from AMP...

Good Evening I just wanted to ask a question here, is it possible to build a static binary that I could sep to a client machine, and have it do it''s first run to connect to a puppet master? My plan is to use a static binary as a sort of installer for particular systems, as I don''t really want to have to install puppet client on 70 systems by hand! These are all production

I have come to realize that I don't have to have a g729a license in order to make use of an ATA-186 or 7460 connecting to another 7460. I just need to allow the codec in sip.conf. Now what ramification does that have when I dial out over one of my analog line (connected to * by a channelbank and a T100P) using my 7460 or ATA-186. The only benefit I am looking for is reduced bandwidth

...mple vpb.conf adtranvofr.conf enum.conf john_todd modules.conf privacy.conf sip.conf z2.conf agents.conf extensions.conf logger.conf musiconhold.conf queues.conf skinny.conf zapata.conf alsa.conf festival.conf manager.conf orig rpt.conf sprackett asterisk.adsi iax.conf meetme.conf oss.conf rtp.conf telcordia-1.adsi asterisk.conf indications.conf mgcp.conf parking.conf s2.conf voicemail.conf [root@liza asterisk]# ------------------------------------------------------------------------ ---- - one other that might have so...

...NS+++++ EXTENSIONS+++++ EXTENSIONS+++++ EXTENSIONS+++++ EXTENSIONS+++++ EXTENSIONS+++++ EXTENSIONS+++++ EXTENSIONS+++++ EXTENSIONS+++++ EXTENSIONS+++++ EXTENSIONS+++++ EXTENSIONS+++++ EXTENSIONS ; Asterisk Management Portal (AMP) ; Copyright (C) 2004 Coalescent Systems Inc ; dialparties.agi (?www.sprackett.com/asterisk/) ; Asterisk::AGI (?asterisk.gnuinter.net/) ; gsm (?www.ibiblio.org/pub/Linux/utils/compre..) ; loligo sounds (?www.loligo.com/asterisk/sounds/) ; mpg123 (?voip-info.org/wiki-Asterisk+config+mus..) ; include extension contexts generated from AMP #include extensions_additional.conf ;...

How many times do you ask for help here before getting a respone? Every single thing I do No matter what I get busy extensions. I am willing to pay someone to help here. Anybody got a clue? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050802/d0d1326c/attachment.htm

Hi, I'm using an external SIP FXO gateway with Asterisk. I've got a fax machine hooked up to an external SIP FXS. I can currently send faxes through asterisk. I've been trying to get fax autodetection to work for inbound pstn calls. The relevant section of my extensions.conf is as follows: [from-pstn] exten => 8000,1,Macro(fixcid) exten => 8000,2,Answer exten =>