similar to: Testing Framework

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, I've just received the following mail: ======================================================================= SETUP Q.931 Message - - CALLED PARTY NUMBER - -- NUMBERING PLAN IDENTIFICATION (Octet 3) - --- 0000 (Unknown) Full Octet = 80'h, you have set to 81'h. TYPE OF NUMBER should also be unknown which I have been told you have

Hi all, I've released another free app for the iPhone and iPod touch - this one lets you read the Daily Asterisk News. Hope you enjoy it :D http://www.venturevoip.com/news.php?rssid=2371 -- Cheers, Matt Riddell Managing Director _______________________________________________ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, Has anyone ever seen an Snom320 lose settings? It's been working fine for months and then I got a call this morning saying that it was asking for country, timezone etc. I logged in remotely, and it had lost the server address, username, password, mailbox and ringtone. - -- Kind Regards, Matt Riddell Director

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Has anyone done any integration with this? All I know so far is that it appears to use some non standard form of SIP. Any pointers? - -- Kind Regards, Matt Riddell Director _______________________________________________ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News -

Hi, I was looking round on the Internet and saw there was no definitive list of free applications available for use with Asterisk, so I thought I'd compile a list for you all. If there's anything that you know of that is actively maintained but not in the list below, let me know (bear in mind I'm not including distros or Asterisk packagings in this list). Hopefully there are a few

Hi, I have these extensions: exten => 101,1,Dial(SIP/101,15) exten => 102,1,Dial(SIP/102,15) exten => 0,1,Dial(SIP/101&SIP/102,15,r) They work fine and I get the ringing sound if I dial them directly. However, I also have this extension: exten => s,1,Answer() exten => s,2,Background(viagenie) exten => s,3,WaitExten() The ringing sound doesn't work for any extension

Hi all, I've uploaded a free app for the iPhone called AsteriskRef to the Apple AppStore. This allows you to lookup applications and functions using your iPhone or iPod touch so you don't have to jump out of extensions.conf or open another terminal tab. It currently supports applications and functions from Asterisk 1.4, but I'm adding 1.6 and trunk at the moment. It currently

Hi, I'm writing a node.js backend to pass events via a websocket to a CRM. Basically what I want to do is notice when things happen (i.e. new channel, new bridge etc) without sending the channels to the Stasis app. The channels I'm interested in are agents who are in a queue only because they are in a realtime MySQL database for the queue_member_table. There doesn't appear to be a

> On 27Apr, 2015, at 16:39, Motty Cruz <motty.cruz at gmail.com> wrote: > > forgot to mentioned I am running Asterisk 1.8.22.0 on CentOS. > > Thanks, > > > On 04/27/2015 02:38 PM, Motty Cruz wrote: >> here is what I have: >> exten => _9XXXXXXX,1,Set(l_HomeAreaCode=381) >> >> exten =>

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi all, anyone else seen RX/TXFax crashing Asterisk on latest Asterisk SVN? I've now seen it on two machines I tried to set up - one it seems because the tiff file was malformed, but the other is doing: tiff -> tx fax -> zaptel -> pstn -> ddi -> zaptel -> rx fax -> tiff The above crashes every time. If no one else has

?I meant how many calls are in progress on a particular trunk. (Sorry - I didn't even think of the other interpretation). ________________________________ From: asterisk-users-bounces at lists.digium.com <asterisk-users-bounces at lists.digium.com> on behalf of Matt Riddell <lists at venturevoip.com> Sent: Sunday, June 28, 2015 9:26 AM To: Asterisk Users List Subject: Re:

Hi all, Just ran into some issue with the originate AMI command. It seems that there is a limit of around 120 calls I can place with the originate command simutanously. By that I mean sending Asterisk a lot of originate command very fast. Anyone know if there is a limitation? Thnx. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Ah, no, you misunderstand. Asterisk wouldn't care one little bit what is on the page - Chromevox would do all that. A screenreader usually tabs or arrows their way about, selecting headings to read content. Thus, Asterisk ONLY needs to be able to hear content FROM the browser and pipe it to the channel, and pass keypresses back TO the browser. The human is the parser, if that makes sense?

Hi, Last night I was working on trying to get Fax for Asterisk installed. I followed all the instructions, but noted that the download selector only provider an X86 32 Bit binary. The machine I'm using is: 2.6.26-1-amd64 #1 SMP Fri Mar 13 17:46:45 UTC 2009 x86_64 GNU/Linux Consequently, on load it presents an error: [Sep 12 10:25:35] WARNING[31306]: loader.c:368 load_dynamic_module:

Hi all, So, we reverted the LibPRI version and tested it, and then tried with the latest version of everything. Still no changes. The BRI line is in PTMP. If we set the configs to PTMP in the genconf_parameters and try it, we get the following: [Jan 21 17:32:20] ERROR[20341]: chan_dahdi.c:12645 dahdi_pri_error: Unable to receive TEI from network! If we set it to PTP (which it is not) we

On Thu, Jun 18, 2015 at 1:26 PM, Matt Riddell <lists at venturevoip.com> wrote: > Did you buy the number from your carrier? Maybe it?s set on their side > for the trunk. > That's what I think too, but they are denying this. I think what's happening is they have a customer service guy interpreting logs (probably incorrectly). When I had a Century Link POTS line, I had a

Hi, I noticed that asterisk manager interface will only accept the originate commands in sequential order. For example, if I want to ring two extensions through the AMI, and while first extension is ringing, AMI won't execute and ring second extension until first extension has answered the call. Anybody has any ideas as I had the same results even tested with telnet commands to AMI interface.

Dear All Can you please do me favor and let me know how I can hide the subs number being displayed on his phone when he goes off hook ? I mean when the subs goes off hook he sees his assigned number on his phone and I need to disable this feature . I don't know from which configuration file this feature is coming so please let me know how can I disable it . Regards H.Motamedi --------------

Hi anyone know where i can find all internatinal numbering plan in csv and for free or small price ? thanks Jpc

Hello All We have developed an application for Home Delivery Restaurants using Asterisk, Java (JSP/ JSF) and MySQL. Here is listing of its features. If someone is interested then we can provide him more details. - POP up window with caller data containing his/her name, address and transactions history. - In case of new customer, Pop up window with blank form to add customer data and