similar to: Trunking through an old Asterisk box.

Hi! I'm trying to set up a SIP trunk so that I can test calls, etc., between a new Asterisk box, and an old 1.4 box. --------------------------------------------------------------------------- New box: root at asterisk1:/etc/asterisk# head -1 sip.conf #include siptrunk.conf siptrunk.conf: [box1] ; All box1 extensions; see extensions.conf type=peer context=adhearsion host=172.17.0.17 ; IP

Hey, all. Just added an analog card to our dual-T1 system... and clearly I'm doing something wrong. Less interested in having the specifics pointed out than in finding out how/why certain things work. So, really, three things: * What the bloody Hell is the difference between dahdi-channels.conf and chan_dahdi.conf? (And who thought it was a good idea to have two files with,

Hi! I've got a Comdial PBX that I would dearly love to replace with an Asterisk box. However, for various reasons, it appears not to be in the cards. Regardless of what management does, or does not, want, our current VM solution -- some Dialogic card with a "KeyVoice" application -- is dying. I'm 90% sure it's hardware. I'd rather shoot myself than replace the

Thanks to the hard work of many people in the Adhearsion community, I am pleased to be able to announce the immediate availability of Adhearsion version 1.0. Since Jay Phillips first began work on the project in 2006 Adhearsion has changed the way developers think about telephony applications. Now with several years of operating experience and multitudes of applications deployed to production,

The Adhearsion team announces the release of Adhearsion version 1.0.1. Adhearsion is an open source Ruby-language framework for creating telephony applications. This update primarily addresses compatibility with newer versions of other software but also adds native support for Bundler to newly created Adhearsion applications. Here are some highlights from the changelog: Handling of new Asterisk

Not strictly an Asterisk question. I've tried to install adhearsion on TWO relatively fresh CentOS 5.x systems, and I get this... [root at localhost rubygems-0.9.5]# gem install adhearsion Bulk updating Gem source index for: http://gems.rubyforge.org ERROR: While executing gem ... (Errno::ENOENT) No such file or directory - /usr/lib/ruby/gems/1.8/gems/adhearsion-0.7.7/bin/ahn The

Hi, I'm giving Adhearsion a try on a Debian Squeeze. I read here (https://github.com/adhearsion/adhearsion/wiki/Getting-Started) that the command "sudo gem install adhearsion" should "automatically add the ahn command to your system". On mine I can't run ahn without specifying full path (/var/lib/gems/1.8/bin/ahn). Did I miss something ? Regards -------------- next

2011/11/25 Olivier <oza_4h07 at yahoo.fr> > > > 2011/11/25 John Knight <john at classiccitytelco.com> > >> Was your PATH variable modified to add /var/lib/gems/1.8/bin perhaps? >> > > No I didn't. > I would have thought that rubygems installation should car of this (adding > installed gems into users paths). > As I'm new to Ruby, I

Docs? Polycom has docs? Where would one find this fabled land of... err I mean Polycom does stating what ftp servers are supported? Doug. -----Original Message----- From: Ken D'Ambrosio [mailto:ken@jots.org] Sent: Tuesday, March 07, 2006 12:12 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re:

Hey, all. I'm seriously thinking about doing the VoIP thing at home. The perfect platform seemed to be the Sheeva "wall wart" (http://www.marvell.com/products/embedded_processors/developer/kirkwood/sheevaplug.jsp). It's a cute little doohicky with USB, SD-card, Ethernet, and runs on an ARM CPU. I'd like to avoid SIP to my provider, just 'cause it's always such a

Hey, all. We're rolling out VoIP, and I'm wondering about PoE recommendations, as we're going to have to replace our current network equipment. My first inclination would be to just plunk down the cash and do a Cisco system, but I'm relatively certain that would get shot down by finance. Any recommendations for a couple-hundred-port solution with VLANs, PoE, and QoS? Don't

Hey, all. Just wondering if there's a GUI out there -- preferably OSS, but I'll take what-have-you -- that a) can run on an Ubuntu/Debian box, and b) allows a receptionist to see what calls are in-process, and forward calls from their phone to somewhere else. Thanks! -Ken -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean.

Hello, we are using Asterisk with Adhearsion as our application server, with another Asterisk box acting as the office PBX, where all office phones are registered. A REFER to transfer calls within the office results in the Adhearsion application call being dropped, because the leg between the PBX and the app server is terminated by the PBX following the REFER. Is there a way to configure

http://downloads.polycom.com/voice/voip/sip_sw_releases_matrix.html Howdy, all. What's the difference between "split" and "combined" firmware, which can be seen at the above link? I've googled to no avail, I'm afraid. Thanks! -Ken -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean.

-- Luca Pradovera luca.pradovera at gmail.com Hello, sorry, I managed to lose the reply amidst the traffic. What we have here is our application server APP with leg A in AsyncAGI in an Adhearsion application, which after some magic dials leg B on the office PBX through a configured peer. Leg B then decides that user C knows more about the subject, and initiates a blind transfer to C?s phone

Hey, all. I'm planning on migrating a W2K3 server to a Linux solution. It needs to be AD-aware, support ACLs, etc. This isn't something I'm doing Right Now(tm), so I can wait a little bit. A couple questions: 1) Are there any known issues with BTRFS? 2) Which version of Samba would be most appropriate for this? 3) AD integration: I've never really done it (with success); any

Hello, as a variation of our issues with Adhearsion calls dropping when an INVITE comes in for a bridged call, I now have a new issue to contend with. Our call is in an AsyncAGI application, and has been bridged to another channel. The provider that supplies the DID sends a polling reINVITE every 15 minutes (it's a documented Metaswitch behavior amongst others). The reINVITE is seen as a new

Hi, all. I'm doing an upgrade from an Asterisk at Home (Asterisk 1.x) system to stock Asterisk 1.4. Everything's working great, except that all the prompts (both stock system prompts on the new system and people's old recorded VM prompts) sound HORRIBLE. Call quality is great, both internal and external. Any idea as to what might have happened? Could I have brought over a config

Hey, all. I quoted a customer about $100 for some cheap SIP phones. I was planning on using the BT-102's, but he called said they look like "Princess phones," and I have to admit that he has a point. Some of the other inexpensive phones look decent, but (for example) the SPA-841's wiki entry says the remote end gets a lot of static. Since it'll be being used from a noisy

> Post your extensions.conf and what's on the CLI (asterisk -r) As requested: # cat /etc/asterisk/extensions.conf [incoming] exten => s,1,Answer() exten => s,n,NoOp(CallerID is ${CALLERID}) exten => s,n,NoOp(DID is ${DNID}) exten => s,n,Background(enter-ext-of-person) exten => 1625,1,Playback(digits/1) exten => 1625,n,Goto(digits/1) exten => i,1,NoOp(CallerID is