similar to: asterisk-users Digest, Vol 42, Issue 51

I'm trying to get call parking to work, but I've run out of things to try. I can place a call between two internal extensions, then on one extension transfer the call to extension 700, and the call gets parked on 701 but I don't hear the extension number when I do the transfer. I can hangup and call 701 and get the call back. Here's what I see: (comments added on lines starting

Hi, Using Asterisk 1.4.17. I'm calling a PHP script through AGI. No matter what my script returns (0 or -1), AGISTATUS always appears to be 0 = SUCCESS. I was wanting my script to be able to return a value to the dialplan and then test AGISTATUS but it looks like I'm going down the wrong path. Any suggestions? Thanks, Brian -------------- next part -------------- An HTML attachment

Hello again, I am using mix monitor and the majority of the sound records perfectly. I then get a "Internal Data Stream Error" near the end of the sound file. Has anyone ever seen this? I am allowing the ULAW amd ALAW codecs and an example dialplan entry is ; ; phone line phone1 exten => phone1,1,Answer() exten => phone1,2,MixMonitor(test.wav|av(0)V(0)) exten =>

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.

Hello, I am wanting to close a specific channel for example; SofthangUp(SIP/EXTEN-UNIQUEID) but the problem is the channel is assigned a unique id as well. The need fits into the idea of receiving a call from a higher status user and thus closing a specific channel to allow the higher priority call to route through the dial plan to the freed extension. Any ideas welcome. Many thanks

Hi All, I have an issue with the ODBC voicemail storage option with asterisk. All appears to work fine, however, I get several sql execute warnings. I was wondering if anyone out there could help me get to the bottom of what is causing this and how I could possibly go about rectifying it. The warning message we are getting is as follows: WARNING[30115]: app_voicemail.c:1280 delete_file: SQL

?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:

this code worked for me, here is what I did and worked for me: exten => 1381+NXXXXXX,1,Set(CALLERID(number)=3817383444) exten => 1+NXXNXXXXXX,2,Dial(SIP/SIP-Provider/${EXTEN:1},80) Thanks for you help! On 04/27/2015 02:56 PM, Matt Riddell wrote: > >> On 27Apr, 2015, at 16:39, Motty Cruz <motty.cruz at gmail.com >> <mailto:motty.cruz at gmail.com>> wrote:

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 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 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

> 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 =>

I've written another round up article on soft phones that you can use with Asterisk. I've tried to omit any that required you to sign up before downloading, or had time limits to usage etc. There are a couple in there that actually have their own servers but may be interesting nonetheless. I also noticed while writing it up there's a version of Firefly for Mac (beta) now.

Hi, We've just completed an interview with John Todd - the Open Source Community Director for Asterisk. Some interesting comments in it - hope you enjoy it: http://www.venturevoip.com/news.php?rssid=2200 -- Cheers, Matt Riddell Director _______________________________________________ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php

> On 18Jun, 2015, at 13:44, Greg Woods <greg at gregandeva.net> wrote: > > I am certain that the old number that is showing up as my caller ID is not present in any of my config files (that includes sip.conf and iax.conf, everything in the /etc/asterisk directory has been checked). Did you buy the number from your carrier? Maybe it?s set on their side for the trunk. -- Cheers,

> At any rate, if I can figure out the right way to set the caller ID explicitly, and assuming they honor it if I do, then none of this will matter. Ok, so just do exten => s,n,Set(CALLERID(all)=?Greg? <5551234>) https://wiki.asterisk.org/wiki/display/AST/Function_CALLERID -- Cheers, Matt Riddell _______________________________________________ http://www.venturevoip.com/news.php

> On 27Jun, 2015, at 15:34, Michelle Dupuis <mdupuis at ocg.ca> wrote: > > Is there a simple way to get call volume from a particular trunk within the dialplan (for conditional branching)? Do you mean large number of calls or how loud the call is? -- Cheers, Matt Riddell _______________________________________________ http://www.venturevoip.com/news.php (Daily Asterisk News)

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

-----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, Does anybody know how to properly execute a transfer (without using the |Tt option) from a GS100? Scenario: 1. I call from X-PRO (ext 1100) to Grandstream (1101). 2. Grandstream answers. Call is established. 3. Press [TRANSFER] on the Grandstream. X-PRO caller is put on hold. Grandstream gets dial tone. 4. Grandstream dials 1103 (the extension of another GS100). 5. Grandstream hangs