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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 Solution) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dia...

...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 _______________________________________________ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss) -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.7 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.or...

Dear list, I am looking for an open source SIP client(or any SDK) that can work on a browser. It may be based html5, javascript, flash, adobe air. I have done some research myself and I would like to ask the community if they have any further hints for me. Real life experience would be awesome. Thanks, Regards, Arstan Jusupov -------------- next part -------------- An HTML attachment was

...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: [asterisk-users] Branch based on call volume On 27Jun, 2015, at 15:34, Michelle Dupuis <mdupuis at ocg.ca<mailto:mdupuis at ocg.ca>> wrote: Is there a simple way to get call volume from a particular trunk...

...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 programs in the list that even the most seasoned Asterisk professionals haven't seen before: http://www.venturevoip.com/news.php?rssid=2184 -- Cheers, Matt Riddell Director _______________________________________________ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems)

...s applications and functions from Asterisk 1.4, but I'm adding 1.6 and trunk at the moment. It currently requires OS3.1.3, but I've got another version under review at the moment which will run on 3.0. Hope you like it, let me know if you have any questions. More info here: http://www.venturevoip.com/news.php?rssid=2353 -- Cheers, Matt Riddell Managing Director _______________________________________________ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dia...

-----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 - html) http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss) -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.7 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.or...

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

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

...n,Dial(SIP/SIP-Provider/${dialnumber},80) >> >> not having success; >> >> "Got SIP reponse 503" Service Unavailable? Can you send us the console output when you make the call? -- Cheers, Matt Riddell _______________________________________________ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/pabx_on_disk.php (PABX on a Disk) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/cc.php (Call Centre Solutions) -------------- next part -------------- An HTML attachment was scrubbed... URL: &lt...

Hi All, I am trying to use the AMD (Answering Machine Detect). But it is not sending the AMD_Status as either the Human or Machine, it hangs up in middle. can any one suggest us, what might be the problem and possible solution to it. below is the log -- Executing AMD("SIP/sip-ffe0", "") in new stack -- AMD: SIP/sip-ffe0 14082284927 (null) (Fmt: 4) Apr 23 08:00:26

...transfer . We tried but it didn't get through on our Asterisk . Can you please let me know what extra config needs to be done for putting it into operation ? Regards H.Motamedi ? ------------------------------ Message: 4 Date: Tue, 01 Sep 2009 17:06:16 +1200 From: Matt Riddell <lists at venturevoip.com> Subject: Re: [asterisk-users] Asterisk Web Meetme module not loading To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com> Message-ID: <4A9CABC8.10209 at venturevoip.com> Content-Type: text/plain; charset=ISO-8859-1; format=flowed On...

...r 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? On 17 October 2016 at 15:22, Matt Riddell (lists) <lists at venturevoip.com> wrote: > This is a really interesting project but I think it's going to be seriously > hard. You're going to need to parse meaning from a site, and that's not an > easy thing to do. > > If you're focused on a few of the bigger sites then it might be easier. &g...

...Asterisk events like you can in the Asterisk manager without polling for channel statuses or sending the channels to the Stasis app and recreating the logic of the Queue application. Is this a correct assumption? -- Cheers, Matt Riddell _______________________________________________ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/pabx_on_disk.php (PABX on a Disk) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/cc.php (Call Centre Solutions) -------------- next part -------------- An HTML attachment was scrubbed... URL: &lt...

G'day all, I'm using Asterisk 1.4 and am trying to work out a way to bring people into a conference call. In the ideal scenario two people would be talking and one of them would push some keys, then a phone number and then the three of them would be in a conference. From there they should be able to bring in other people as well. This seems to be what the Asterisk n-way call HOWTO

How to make outgoing calls from DID and what is theway to get incoming calls from DID. -- ----- Thanks and regards Virendra Bhati +91-9172341457 Asterisk Engineer -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110526/3f19091d/attachment.htm>

>>I'm not sure that your question is clear. You'll probably want to be more specific. >> What is pulse? You mention "as a user", are you talking about voicepulse.com ? >> What are you trying to do with pulse? >> What problem are you running into? Sorry Rusty... I am trying to get Asterisk 11 to co-exist with a centos 7 box that has pulse audio running as

>Soumya Kat wrote: > What I would like to know is how to get information such as SIP users, > number of SIP connections and traffic associated with those from asterisk > using a C Code. >Russell Bryant > There is actually no good way to do this inside of Asterisk right now. It's > certainly all possible ... it's just software ... but there is no > straightforward

We just completed a new implementation of voicemail for Asterisk. It's much cleaner than Comedian mail and can emulate several voicemail user interfaces, including Audix. It's a great replacement for Audix. All of the sounds/prompts are presently being re-recorded by a professional female voice. If you are interest in the app, let us know at nt_jnewman at yahoo.com. Justin

...00200 plen:34) 1191205421.801091696: 00 81 00 00 08 01 01 05 04 03 80 90 a3 18 01 83 6c 09 01 c3 38 37 32 38 34 35 30 70 05 81 36 35 30 30 This is driving me crazy. Is there anyone who can help? - -- 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 - html) http://feeds.venturevoip.com/AsteriskNews (Daily Asterisk News - rss) -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.7 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.or...