search for: anselm

please can anyone help me knowing if i can install Linux and Asterisk on HP servers _________________________________________________________________ Put your friends on the big screen with Windows Vista? + Windows Live?. http://www.microsoft.com/windows/shop/specialoffers.mspx?ocid=TXT_TAGLM_CPC_MediaCtr_bigscreen_012008 -------------- next part -------------- An HTML attachment was

Hello ppl, Any way to specify alphabetical exten patterns in the dialplans on Asterisk? All my users would have alpha/numerical ids. I don't want to add a line for every user in my dialplans. I searched around, but couldn't get anything useful. Any way to get around this? Thanks in advance - Benjamin Jacob. EMAIL DISCLAIMER : This email and any files transmitted with it are

...tialise end end ... success = MySuccess.new success.expects(:body).returns("this is the body") .... Which is a little bit frustrating - it''s to avoid having to do this type of code I''m using Mocha in the first place ! Any ideas ? Thanks for this great library btw!, Anselm [1] http://ruby-doc.org/stdlib/libdoc/net/http/rdoc/classes/Net/HTTP.html -- ------------------------------ Netuxo Ltd a workers'' co-operative providing low-cost IT solutions for peace, environmental and social justice groups and the radical NGO sector Registered as a company in Englan...

Hi! I am getting inbound caller ID fine bout not out. I am in Sweden and suing Rixtelcom /POrt80 as provider. anyone knowing what is wrong? Also is anyone knowing about Swedish voices to trixbox/Asterisk? I have male now and am looking fro female voices. Regards Mattias -- Mattias Andersson -------------------------------- Storskiftesv?gen 6 145 60 Norsborg m. +46-70-799 44 41 h. +46-8-641 38

Dear all I have FAX machine connected with audiocode SIP device i am trying to send fax and when negosiation going on and i start send fax button then my after half page it got stuck in fax machine so is there any codec problem i am useing ulaw/alaw is it fine or not anybody have idea about sending fax with SIP connected device --------------------------------- Ready

Dear all I have configure asterisk with 100 SIP PHONE ( SNOM ) but now thing is that my boss need phonebook feature find extention number by Pbook so i have read about it there is a feature in asterisk but it is with voicemail now i have IP SIP phone of SNOM so how to fine phone number by SIP phone ?? how to asterisk directory work ? Rgd satish patel

...ease with asterisk on the LAN (which would make those phones worthless for me...)? My sip.conf relevant parts are [sip505] mailbox=05 callerid=505 type=friend username=sip505 secret=abcd123 context=sipclient host=dynamic nat=yes disallow=all allow=alaw allow=gsm allow=ulaw Thanks for all input, Anselm

Is what I am trying to do in this context possible. That is changing the incoming CALLERIDNUM. In this case if the incoming CALLERIDNUM is not preceeded by a "1" I want to add a "1". Often calls come in without the preceeding "1" and this plays havoc with my redial if the 3 digit area code matches a local 3 digit extension. All my outside calls are 10 digits or 1+10

Hi group, I'm trying to configure the email notification when a user leave a voicemail, but don't work (send email notification). I configured esmtp in my linux box, if a try to use it with command line, it works fine. (echo "Hello" | sendmail a@b.com -f b@c.com). My voicemail.conf [general] format=wav49 attach=yes serveremail=anonymous@abc.com fromstring=Asterisk

Sorry if you see this message repeated twice. I would like to set up hard phones and softphones with the same extension so that when anybody in the company dials an extension, each user's hardphone and softphone will ring at the same time. I've tried setting this up before, but I noticed that the last sip device to register with the same extension is the only one that rings when the

Hello Now that I received an OpenVox PCI card (www.openvox.com.cn/products_detail.php?genre_id=9&id=28), I'm ready to try and set up a voice server with Asterisk. We need the following features: 1. When customers call in, they should hear a voice menu asking them which software they're calling about 2. Next, they should be able to leave a voice message to explain what their problem

All, If I have video phones behind an asterisk server (with 2 network cards) and all the phones have extensions. Internally everything works great. Now for people that want to call my video phones external to my office is there a way to do that? On the extenal persons phone enter an IP/EXTEN where IP is my server and not the phone? Can that work? Would I have to have PUBLIC IP address for every

I am trying to set up a Conference room where users are put on hold until the host arrives. I have figured out that the A option activates marked mode and the w option is used to activate the waiting until the marked user arrives. This seems to be what I need. What I can't seem to find is how do I mark a user? Thanks _____________________ Kevin Savoy Business Unit Telecom Analyst 2218 4th

...ers] MeetMe Conferencing and Marked Mode I'll give this a try but seems silly to require 2 different extensions for one conference room. Thanks for the input. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Anselm Martin Hoffmeister Sent: Wednesday, December 13, 2006 7:16 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] MeetMe Conferencing and Marked Mode Am Dienstag, den 12.12.2006, 13:08 -0600 schrieb Savoy, Kevin - Williston, ND: > I am trying to set up a Co...

VMukti.com ----- Original Message ----- From: "Anselm Martin Hoffmeister" To: "Asterisk Users Mailing List - Non-Commercial Discussion" Subject: Re: [asterisk-users] Filesharing + video + voice supported Soft phone Date: Mon, 17 Sep 2007 15:06:03 +0200 Am Montag, den 17.09.2007, 05:09 -0700 schrieb satish patel: > Dear...

...the process terminated correctly. It seems not to be possible to assign $? directly... For now I do this before my test : `ls > /dev/null` Which ensure that $? is zero ; but this is not a very nice way of doing it ! If anyone here has a good trick for this, it will be very welcome :) Thanks, Anselm -- ------------------------------ Netuxo Ltd a workers'' co-operative providing low-cost IT solutions for peace, environmental and social justice groups and the radical NGO sector Registered as a company in England and Wales. No 4798478 Registered office: 5 Caledonian Road, London N1 9DY...

...rs] Filesharing + video + voice supported Soft phone Date: Mon, 17 Sep 2007 16:57:25 +0300 There is for sure X-lite and other similar but you won't get file sharing which is meaningless either way. If you need it then use msn :-) . http://www.counterpath.com/xlitedownload.html Anselm Martin Hoffmeister wrote: Am Montag, den 17.09.2007, 05:09 -0700 schrieb satish patel: Dear all I have setup of asterisk 1.4.11 Now i want soft phone which one support file sharring + video + voice call with asterisk SIP is there any so...

...ing, mock object and stubs - Very quick introduction to mock objects and stubs Example with Mocha - The core of the guide ; shows how to use Mocha with a continuing example The Mocha library - How the library is structured Mocha tips - Some tips I would''ve found useful at first :) Best, Anselm -- ------------------------------ Netuxo Ltd a workers'' co-operative providing low-cost IT solutions for peace, environmental and social justice groups and the radical NGO sector Registered as a company in England and Wales. No 4798478 Registered office: 5 Caledonian Road, London N1 9DY...

When i send more than one messages shortly after the other, my log (/var/spool/asterisk/sms ) looks like this and only two of four messages arrive. What am i doing wrong ? I am using an AVM B1 PCI with chan-capi and 1.4.4. and also, when sending with smsq -x only two of the messages are handled. (i thought, asterisk itself handles the queues ? ) Here the log: 2007-07-09T15:04:14 YOM04 0 -

Hi all. We have Snom phones which do have a defined key in order to drop incoming call WITHOUT answering. Pressing that key, a "SIP/2.0 486 Busy Here" message is sent back. We have other phones (I.E. DECT Siemens C450IP, or ATCOM 320 or other) which DO NOT have any key to do that (or the key does not work, as is with Siemens C450 IP ): you have to answer and immediatly after hangup the