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Hi all, I'm seeing a lot of these messages: [Apr 30 20:28:57] WARNING[5402]: chan_sip.c:12543 handle_response: Remote host can't match request NOTIFY to call '2e4a02807750b7024d5ff09c668fa0f7 at 10.0.0.2'. Giving up. [Apr 30 20:28:57] WARNING[5402]: chan_sip.c:12543 handle_response: Remote host can't match request NOTIFY to call '0755ad8f40b9d09d491b635e70bb8905 at

Hi, I know this is a little OT but there are many Asterisk users of the excellent Siemens DECT/VOIP phones like the S685IP and 475IP and this is probably newsworthy for them. One of the biggest bug bears has been no mute function on the handset. When I woke up this morning, the handset told me there was a firmware update. I updated and then visited the web site to find out what had been

Sorry if this appears twice - I sent it yesterday and it hasn't yet arrived so trying again: ====== Hi all, I have a Polycom C100 communicator and from what I can find about it, the real work done to make the audio as good as it is, is done in the software they install on Windows XP. It is a USB connected desktop handsfree mic/speaker device and it does work on Linux O.K. but I get some

Hi all, This isn't meant to be spam I thought some of you might find it interesting. Packt Publishing approached me a few weeks ago and asked if I would like to review a book or two for them on my blog. The first one they sent me is called Asterisk Gateway Interface Programming and has only just been released. It was written by Nir Simionovich. You can read my review here:

Hi all, Can some one suggest me an IAX client for Linux and Windows? I used KIAX once, but know it seems complicated to have it working on Ubuntu. Thanks. Ronaldo.

Hi all, I built my first asterisk using the traditional (?) .conf files and constructs. I recall reading books at the time about AEL but it seemed "new" and untested so I left it alone. Now, I'm interested to poll the audience here to see if I should look into using AEL instead (or in addition to) for future work. TIA

Hi, I have three Snom M3s at the moment but getting pretty fed up with the issues :( I am UK based and would be interested to hear of other peoples recommendations. Key features :- * VM Notification * Good Range * G729 codec support * Common/Private Address Books per Handset(s) TIA, Best Regards, -- This message has been scanned for viruses and dangerous content and is believed to be

I lurk and comment a little on here and have been playing with * for a short while. I am interested in hearing about the pros and cons for using a database backend to Asterisk. My current setup is simple, out of the box with config files in /etc/asterisk and logs etc going into /var. I notice a great many of the contributors here seem to use a db backend (is this also called Real Time

http://gigaom.com/2010/08/03/2600hz-project/ -- The Open Learning Centre http://www.theopenlearningcentre.com

Hi all, I am trying to understand how I can get a simple IVR scenario to work properly (having already removed most of my hair...). The basic requirement is as follows: * Caller arrives at our main number * Caller is greeted and then told they can enter an extension number, if known, or wait and their call will be connected to an available rep. * The IVR then dials a group of extensions (if

Hi, I know the answer is probably negative, but there's no harm in asking right? I have a Polycom C100 communicator and from what I can find about it, the real work done to make the audio as good as it is, is done in the software they install on Windows XP. It is a USB connected desktop handsfree mic/speaker device and it does work on Linux O.K. but I get some problems with echo when

Can I do this? I have a x100p card on my PSTN line and I have an incoming context for these calls which uses the "s" extension. I'm wanting to set up a simple IVR and would like to be able to test the dialplan as I go. But having to dial-in on my PSTN line each time is going to cost me money. Can I connect to my zap_incoming context from my locally connected SIP phone? I'm

I'm trying to get emailing of voicemail messages to work and by and large it does... However one email address is quite long in comparison to others I am testing and it fails to get delivered. For example - this one works and gets delivered: [Nov 5 18:35:14] DEBUG[2509]: app_voicemail.c:1957 sendmail: Sent mail to ****.****@***.*****.com with command '/usr/sbin/sendmail -t' And

Hi All, I've just upgraded our CRM and it has an Asterisk Integration Module that I would like to test out. The CRM is running on one of our hosted servers in the cloud. The Asterisk server is running in my office. I am running Asterisk 1.4.21.2~dfsg-1ubuntu3. Reading the page http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20manager.conf got me a little concerned

Hi all, I'm sure this has been done before but I just can't figure it out. On my * box I have a simple IVR: [tolc_menu] ; Welcome and information to callers exten => s,1,Answer() exten => s,n,Wait(2) exten => s,n,Background(welcome-to-tolc) ; Say Hello exten => s,n,Wait(1) exten => s,n(tryagain),Background(enter-ext-of-person&or) ; Enter extension number if known, or

When you configure the Siemens gigaset handsets (I have S685IP), there is a single option for all handsets to use either the POTS interface or VOIP as the default outbound destination - you then need to add a dial suffix if you want to use an alternate outbound route. Does anyone have any suggestions as to how to make just *one* of the DECT handsets only use the POTS but others default to

Hi all, I am close to purchasing some new DECT phones for our home office here in the UK. We use Asterisk and I am sorely tempted by the Siemens C475IP or the "soon-to-become-available-in-the-uk" S685IP. Both systems have great feature sets and, on-paper at least, look to be the bee's knees. Anyone got any skeletons on them? Thanks Alan -- The way out is open!

Hi, I'm building a test asterisk server and building the latest kernel I got to wonder if there are any specific recommendations about schedulers and so forth for optimum performance. There are a few areas that raise questions in my mind and I wonder if anyone has any opinions/comments on which settings are most suitable for use with asterisk: SLAB Allocator (SLAB or SLUB?) Tickless

Hi, am building the latest version of Asterisk (1.4.13) on a self-build Linux host (based on LFS-6.3). Version 1.4.12 built installed and worked fine. Last night I upgraded the kernel to 2.6.23 and rebuilt the zaptel driver package 1.4.5.1 against it. That seemed to build and install O.K. too. I dl'd the 1.4.13 source tarball and tried to build that: ./configure ran O.k. 'make

I wonder if anyone has any ideas on this. I have recently migrated my server from a custom built Linux with Asterisk, Zaptel and Oslec running fine, to Ubuntu Server 8.10. I have Asterisk installed via synaptic at it works fine. I have built and installed the zaptel package by doing the following commands: sudo m-a -t build zaptel cd /usr/src sudo dpkg -i zaptel-modules-{version}.deb sudo