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Asterisk Users Mailing List - Non-Commercial Discussion
January 2008 - page 1
Polycom Digit Map (5 Replies)
Michael Munger — 119914653301 Jan 2008*
That was one of the many iterations I tried already. It seems to respond in that it recognizes that I am dialing 01186106887XXXX, but then it only con...
Problem with Polycom Soundpoint IP 320 Hardphone (4 Replies)
Glenn Gillen — 119919672001 Jan 2008*
Hey all, I've setup my asterisk install on a CentOS5 server, I've got a few IAX2 and SIP softphone clients connected on the same subnet and at...
satish patel — 119920191401 Jan 2008
Dear all I have cisco phone 7974 i have useing SIP protocol to register phone on Asterisk and it is working fine but i have one problem when how do i ...
(no subject) (2 Replies)
lists65 — 119921434901 Jan 2008*
[ArchiveOrange]: There doesn't seem to be anything here...
John Miloo — 119921850701 Jan 2008*
Hello Comunity, How can I get Asterisk realtime working with Postgres? (without ODBC)? Thanks John /doc/realtime.txt in Version 1.4 Beta2 Currently th...
Password protect a queue from callers? (1 Reply)
Caza Henha — 119924549202 Jan 2008*
Hi, We currently testing a trixbox/asterisk installation and have used Freepbx to set-up and configure the box and it is running tremendously well. We...
Gotoiftime help (6 Replies)
troxlinux — 119924724802 Jan 2008*
hello list, I am trying to arm an ivr for schedule of office and outside of office [general] static=yes writeprotect=no autofallthrough=yes cleargloba...
Mike Dent — 119925345102 Jan 2008*
Hi, just wondered if it was the same firmware on both devices? thanks Mike...
Trixbox and mail2fax (3 Replies)
Daniel — 119925997002 Jan 2008*
Hi there, is there any howto how do i configure a asterisk/trixbox for mail2fax? The fax must be send over sipgate or other SIP peers. (i dont have an...
Vieri — 119928110502 Jan 2008*
Hi, Before I report a bug on http://bugs.digium.com, I would like to know if someone is seeing the same error message. Personally I am not using wctdm...
Adam Moffett — 119928337302 Jan 2008*
I asked this question last week and never got an answer. I also didn't find the answer in the wiki. I think it would be nice if asterisk would che...
One Way Delay in Audio Over Analog (3 Replies)
Brian Alexander — 119928417102 Jan 2008*
I have been trying to track down the cause/fix for a problem and I am out of ideas... I am hoping one of you can point me in the right direction. The ...
Lamps on Snom phones (1 Reply)
Phil Knighton — 119928475902 Jan 2008*
HelloHappy New Year to all!!I've just completed porting from Asterisk 1.2 to 1.4. I did this by doing a clean install on a new box, and moving ove...
Invalid extensions (4 Replies)
Gilberto Nunes Ferreira — 119929112602 Jan 2008*
Hi all First I want to wish for everone a happy new year... Well... I have run asterisk 1.4.16.1 in a server. I have this IVR, in extensions.conf: [ur...
bilal ghayyad — 119929122002 Jan 2008*
Hi Rob; Big thanks for your kindly help and answer, so rtp.conf file is used by sip and h323 only, correct? In that case if I am going to use the sip ...
AGI stream file (3 Replies)
Timothy Legge — 119929751202 Jan 2008*
Hi I have created a rudimentary perl script that does most of what I want but occasionally in seems that a file will not play. I see the message getti...
Incoming Calls (9 Replies)
Paulo Pinheiro — 119929970202 Jan 2008*
I am having a problem that I would like to verify if someone could help...I am using bandwith.com as my SIP TRUNK provider. When I place the phone num...
Brian McManus — 119930129002 Jan 2008
This is off topic but Asterisk related, if any of you asterisk users also have an IPTV deployment or IPTV middleware, toroid, my new open source middl...
How to stop the update of astdb? (1 Reply)
Aryanto Rachmad — 119930576202 Jan 2008*
Hello everybody, I am not using astdb (no func_db and app_db) so I am wondering why asterisk is always updating it. The interval of the update is not ...
The Asterisk Development Team — 119930994402 Jan 2008
The Asterisk.org development team has released Asterisk version 1.4.17. This release contains a fix for a SIP security issue, as well as a number of o...
Asterisk Security Team — 119931105702 Jan 2008
Asterisk Project Security Advisory - AST-2008-001| | important to note that a dialog must have already been || http://downloads.di...
is Power fail transfer possible with asterisk? (4 Replies)
John covici — 119932067603 Jan 2008*
Hi. I have a client who wants some way that his analog phones can call out even after the power is out and the UPS has died -- some way that a phone c...
Douglas Garstang — 119932157803 Jan 2008
When I saw the subject I thought the poster was maybe asking if was possible to transfer the live RTP stream from one Asterisk system to another in th...
Asterisk E1/T1 Card configuration (1 Reply)
Jeng Yu — 119932498603 Jan 2008*
My Esteemed Gurus! Still learning... I need to read about and learn how to configure Asterisk box here in the lab for digital cards. I'm about to ...
PRI Crapping Out Regularly (1 Reply)
George Pajari — 119932567803 Jan 2008*
We have a server with a TE120 on a partial PRI trunk that several times a day declares the PRI trunk down and stops handling calls until the asterisk ...
Goke Aruna — 119932602603 Jan 2008*
I have used asterisk-1.4.14, zaptel-1.4.7, chan_ss7-1.0.0 on FC7 all went okay. using sangoma a104dx on both machine. I followed the write up on http:...
sip.conf & realtime (1 Reply)
hugolivude — 119932619003 Jan 2008*
Hi - I'm looking into realtime and I'm having a bit of a problem with the SIP part. My review of the posts seems to indicate that I should use...
Snom 370 buton Recordings (1 Reply)
voip crazy — 119932720703 Jan 2008*
Hello all, I am using the Snom 370 phone with firmware Snom370-SIP 7.0.17* *connected to an asterisk 1.2.14 and I can't record any calls using the...
Softphone to be installed on the Mobile (1 Reply)
bilal ghayyad — 119932738903 Jan 2008*
Thanks a lot for the help. But if Asterisk has private IP address and the only way to access it from remote sites is to have vpn connection to the sit...
Christian Stredicke — 119933072603 Jan 2008
BTW I would recommend to move to 7.1.30, this is much better than 7.0.17. CS...
[1.4 + FreeBSD 6.2] Playing WAV PCM file? (16 Replies)
Vincent — 119933442603 Jan 2008*
Hello Happy New Year! I succesfully installed the Ports of Zaptel BSD 1.4.0 and Asterisk 1.4.13 (that's the latest in the Ports). To save CPU, I...
auto dial and IVR (2 Replies)
Rilawich Ango — 119934007403 Jan 2008*
Hi, Is it possible to let asterisk auto dial out and play the IVR? How? i.e. -asterisk auto dial out (use outgoing folder?) -user pick the call -play ...
How does Asterisk scale to 500-1000 phones? (2 Replies)
Jesse Molina — 119934596803 Jan 2008*
Anyone have opinions on how well Asterisk scales to 500-1000 stations, in regards to reliability, system performance, as well as ease of management? A...
Asterisk dialplan date and time operations (8 Replies)
Erik Wartusch — 119935096903 Jan 2008*
Hi all, Im using Asterisk 1.4.11 and I want to proceed some time and date operations in my dial plan. (for a time shifted callback). Should look like:...
Anthony Chapellier — 119935749703 Jan 2008*
Hi, I'd like to know if it's possible to configure Asterisk to automaticaly close calls when the BYE request hasn't been sent by any clien...
Asterisk 1.4 Fax (37 Replies)
Al lists — 119936426703 Jan 2008*
what method is preferred: haylafax and Iaxmodem or spnadsp for faxing....
Polycom VLAN (3 Replies)
Jeremy Mann — 119936500303 Jan 2008*
Just curious, if I have my Polycom IP 550 phone VLAN tag 30, will the packets I send from my PC(on the PC port of the phone) have the same VLAN tag? T...
Missing "zap" command in Asterisk 1.4.16 (7 Replies)
Raúl Gómez C. — 119937049603 Jan 2008*
Hi list, I've just compiled and installed Asterisk 1.4.16 and when I try to run "zap show" I get the message "*No such command ...
Recorded calls skipping (6 Replies)
Jay Moore — 119938344903 Jan 2008*
Greetings, List. I'm having a problem where my recorded calls are skipping every 4-5 seconds are so. I can hear the caller (or callee) just fine a...
Bad Link on Website... (1 Reply)
Stuart Sheldon — 119938397703 Jan 2008*
Not sure where to report this... http://www.asterisk.org/downloads Right hand download box, Asterisk 1.4.17 points to 1.4.1 Just a heads up. Stu...
William Herrera — 119940332003 Jan 2008*
I used to work for "Telefonica of Puerto Rico" installing Asterisk, so I have installed few of them. I installed one last week (downloaded a...
Gregory Malsack — 119940686004 Jan 2008
Yea, sounds like they've planned for this issue. Kevin, is there an sdk that can be used to create our own binaries should we want to add modular ...
Doug — 119942674104 Jan 2008*
Would this be a firewall problem? chan_sip.c handle_request_register: Registration from sip failed for ACL error (permit/deny)...
A thougt (2 Replies)
Fredrik Söderlund — 119943188704 Jan 2008*
Is there any possibilletys to klick on a telephone nr an it will dail like the case in a mail program if you klick a url://a.b.se it opens a browser a...
Right timing for a queue call (3 Replies)
Andrea Spadaccini — 119944212804 Jan 2008*
Hello everybody, I'd like to have more detailed records for calls related to queues. For instance, if A enters in queue X, waits for Y secs and th...
Digium Asterisk Appliance voicemail & logs (7 Replies)
Gregory Malsack — 119944817704 Jan 2008*
Here is some information I received from my account rep at Digium regarding this information: -- Digium -- That's news to me as well as the rest o...
OT - GEOPRIV and location based SIP services (8 Replies)
Olivier — 119945234504 Jan 2008*
Hi, I'm wondering whether or not it is achievable to build a web based click2dial application that could automatically detect that a user is conne...
Agents and AddQueueMember (4 Replies)
Rajkumar S — 119945794004 Jan 2008*
Hi, I have callcenter running with v 1.2 with AgentCallbackLogin and now trying to move to 1.4 using the example doc, doc/queues-with-callback-members...
Tomasz Zieleniewski — 119945882204 Jan 2008*
Hi, I have the following problem that when asterisk receives SIP response 302 it cannot forward the call I get such debug: [Jan 4 10:43:27] WARNING[18...
Robert Moskowitz — 119946050204 Jan 2008
I have a SIP trunk to Broadvoice. My Asterisk box (1.4.13) is on public addresses behind a firewall. Originally it was behind a Linksys WRT54G running...