similar to: Call routing between two Asterisk boxes using SIP not working ...

Displaying 20 results from an estimated 200 matches similar to: "Call routing between two Asterisk boxes using SIP not working ..."

2009 Aug 26
4
Multiple user registration ...
Hello there! We are planning to use Asterisk on our VoIP platform, and we are spending some brains on a way to provide the following facility: let some SIP user (extension) registrate with more than one client (ATA, SoftPhone, VoipCelular, etc) - what isn't a problem at all -, initiate calls from any of this devices that are registrated with the same user - no problems on tests too -,
2009 Aug 18
2
Platform decision ...
Hello there! During some research on Internet I found the following comparison on site Voip-Info (see, "http://www.voip-info.org/wiki/view/OpenPBX.org+FAQ"): The main points listed on Asterisk's "CONS" that concerned me were: * Conferencing on Asterisk depends on Zaptel hardware and/or kernel modules for timing; * Lack of built-in STUN support for SIP NAT
2009 Sep 02
2
Does L(x:y:z) "Dial" option work on Asterisk version 1.4 ?
Hello there! I'm testing "Dial" call limit option on Asterisk version 1.4.26, but it's not working. The issued command is: "Dial(SIP/${EXTEN}|20|RtT|L(300000:60000:20000))". Am I missing something ? Does it only work with Asterisk version 1.6.X ? Thanks and best regards, -- __At.,
2009 Aug 25
1
Realtime with "rtcachefriends=no" problems...
Hello there! I was testing Asterisk for the last two weeks using the Realtime driver for MySQL, and leaving "rtcachefriends=yes" configured to enable MWI. Today I started making additional tests with "rtcachefriends=no" because we will probably need to use Asterisk without this cache. For some strange reason, calls stop to get routed between the SIP clients. I've
2009 Sep 28
1
How to get "Call-ID" SIP header outside "chan_sip" scope ...
Hello there! I'm working on some modifications on Asterisk to adapt it to our needs considering some particular demandings of the infraestructure we want to provide. Two of these modifications are: 1- A proprietary configuration driver that will communicate with a server that will be the source of information for the entire infraestructure; and, 2- A call control application that will be
2005 Jul 25
3
Wengo config and G729(a)
Hi list! Again Wengo has made changes to their servers that require modifications to * configs. Is there anyone that has the 'new' wengo working with asterisk that could post their configs? Also they switched codecs, now G720a is required to connect. I can only find an (open) G729 codec, is this the same as G729a? Thanks!
2005 Mar 20
2
Echo after upgrade * 1.05 -> 1.06
Hi list! I have a strange echo problem. Two days ago I setup * 1.0.6. at a friend of mine. Just an * server and for outbound calls wengo.fr was used to place calls via sip. He had a strange echo on the line I didn't experience on my setup. Today I upgraded my asterisk 1.0.5 to 1.0.6 and suddenly I have an echo too on sip calls thru wengo!! I already verified wengo was not the source of
2008 Mar 23
1
Storing voicemail in mysql
Dear friends, Asterisk's voicemail functions work fine for me, but I am having difficulty storing the voice messages inside mysql. My real-time CDR recording works so I assume the odbc connection is fine. The voicemail.conf I have is : [general] format = wav attach = yes dbuser=root dbpass=sqlpass dbhost=localhost dbname=asterisk odbcstorage=asterisk odbctable=voicemessages Asterisk shows
2005 May 05
2
Did nufone change allowed codecs?
Hi, I've been using nufone DIDs for months with no problem. Now there are codec problems that prevent any kind of calls working. For example, May 5 13:04:12 WARNING[928]: channel.c:2115 ast_channel_make_compatible: No path to translate from IAX2/NuFone@NuFone/25(256) to SIP/wengo-out-968a(4) May 5 13:04:12 WARNING[928]: app_dial.c:1006 dial_exec: Had to drop call because I couldn't
2006 Oct 11
10
GPL Softphones
Hi, I'm searching for GPLed softphones. I found WengoPhone but actually not available for Asterisk PBX, only for Wengo network. I found Kiax but only for IAX protocol. Did you know a good GPLed softphones which works on Windows ? Thanks Greg -------------- next part -------------- An HTML attachment was scrubbed... URL:
2005 Jan 08
7
France has their (first?) SIP carrier with "unlimited" calls for 6eu/mo
Asterisk must have a reasonably large community here in France judging from the number of people who came out to meet Mark. Either that or we were ALL there :) Something I've been waiting for, a voIP carrier on the models we are used to (low monthly or pay as you go, web account) has just set up their first beta test for 1 euro for the first month, 6euros if you decide to keep it. The basic
2008 Feb 07
6
Asterisk G722
Hello, I have some problems to use G722, when my client sent an invite request to asterisk using G722/16000 codec asterisk respond with G722/8000 codec. I dont know exactly if Asterisk supports G722/16000 codec?? If yes how can I activate It?? Thanks. Rachid. Below wireshak trace:
2007 Oct 02
3
Logwatch for postfix
On CentOS5 with the latest updates applied, the logwatch filter for postfix returns way too many lines from the log. I get an "unmatched entries" message and all messages that have gone through the system is listed. Here is an example: 8F930A8092: to=<morten at foo.bar>, orig_to=<morten at localhost>, relay=local, delay=0.19, delays=0.06/0.01/0/0.12, dsn=2.0.0,
2006 Jun 14
0
Asterisk & wengophone
Hi I use Asterisk with some SIP phone (grandstrea), while with my notebook when I'm out of home/office I use X-lite and all work. Some days ago I try to install wengophone and I decided that I want replace X-lite for use wengophone as client for my Asterisk. One of the reasons is that wengophone support g729 codec. I make some test and I see that is possible to configure other sip server
2005 Jul 04
1
OT : Wengo sucks
Would just like to warn everybody for Wengo.fr Once you sign up there is no possibility to remove your credit card and even though you send them resignation letters they keep charging your credit card. Now I understand what they mean when they say `unlimited subscription'.
2005 Jan 26
1
IAX/SIP Softphone with G729
Subject says it all. We are looking for an open source IAX or SIP softphone that can handle multiple lines and have G729 abilities. We don't mind paying a few bucks for the 729 license. We need open source so we can co-brand. Right now our only solution is to buy $5000 worth of XTen Pro; then we get co-branding and 729. But if we can save $5000, all the better. -Matthew
2006 May 24
2
Video SIP Softset
Sorry if this shows twice but it appears my first message was quarantined because of my digital signature. All, I have been tasked with setting up video conferencing utilizing asterisk. One of the requirements is a softset that has video capabilities. Eyebeam looks promising but I was just wondering if anyone out there knows of any freeware with comparable features of Eyebeam that they
2012 Apr 12
1
Support for R in highlight.js
Hello, I'm the maintainer of a syntax highlighting tool highlight.js[1]. Recently the Kaggle project has announced they wanted to sponsor the development of the R highlighting definition for it[2]. I wanted to drop a line about it here since I suspect this list has much more R programmers on it than our small discussion group :-). So if you're interested and don't mind touching a
2010 Aug 03
10
Returning last value
Hi. I have the following controller of the update action: [code] def update @projeto = Projeto.find(params[:id]) if (@projeto.update_attributes(params[:projeto])) ## Mesmo raciocĂ­nio utilizado no create. @permissaoA = Permissao.find(:first, :conditions => ["usuario_id = ?", @projeto.responsavel.to_i]) @permissaoAvancado1 = Permissao.find(:first, :conditions
2010 Nov 15
4
Best way to connect to a MySQL Database
Is this command the best way to access a MySQL database - MYSQL(Connect connid dhhost dbuser dbpass dbname) ? I thought I heard that using ODBC was a bit more stable. Anyone have any experience? Thanks, Matt