Re: [asterisk-users] Joining an astablished call

-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of neo haux Sent: Monday, 6 May 2013 1:34 p.m. To: asterisk-users@lists.digium.com Subject: [asterisk-users] Joining an astablished call Hi, I don't

Re: [asterisk-users] Load Balancing

I came accross this article (Asterisk rtp mprovements http://www.voip-forum.com/opensource/2013-04/asterisk-rtp-improvements/) mentioning DNS based load balancing. I will give Opensips loadbalance module further reading to better understand how it works Thanks for the tip. 2013/4/25

[asterisk-users] OT - Differences between Aastra 6730i and 6750i series

Hi, What are the main differences between Aastra SIP phones 6730i and 6750i series ? Aastra corporate web site mentions : The Aastra 6730i Series offers exceptional features and flexibility in an open-standard enterprise grade IP telephone for one The Aastra 6750i Series offers features and

[asterisk-users] OT - Question on Aastra 6735i - Was: Sip phone displaying caller name while on call

Hi, 2013/4/19 Olivier oza_4...@yahoo.fr Hello, I've just realized that several phones display both caller name and number while ringing but when on call, caller name is not displayed anymore. Could you recommend a sip phone that still displays caller name during phone call ? Regards I've

[asterisk-users] MRCPSynth() change voice

Hello, I'm trying to change the voice during a spoken text: exten = _X.,1,Answer exten = _X.,n,MRCPSynth(Hello, my name is Daniel. I have a Dutch companion. ###\voice=Xander\ Hallo, mijn naam is Xander.,p=defaultl=en-GB) exten = _X.,n,Verbose(1, ${SYNTHSTATUS}) exten = _X.,n,Hangup This exact

Re: [asterisk-users] Testing 911 call

Quite a few SIP providers will have 911 testing functionality. Our main 911 provider lets you dial 933. Than they read back to you the address information that is transmitted with the 911 call. -- Ringfree Communications David Wessell 828-575-0030 x101 From: James Miller

Re: [asterisk-users] OT - Differences between Aastra 6730i and 6750i series

- Original Message - From: Olivier oza_4...@yahoo.fr Hi, What are the main differences between Aastra SIP phones 6730i and 6750i series ? Aastra corporate web site mentions : The Aastra 6730i Series offers exceptional features and flexibility in an open-standard enterprise

Re: [asterisk-users] My new Polycom 450's can't xfer to 4-digit extension

On 05/04/2013 08:43 PM, Mike Diehl wrote: Hi all. I just installed bunch of IP450's and everything went well and my customer is happy except that they are unable to transfer calls to other extenstions. They can dial them directly just fine. However, when the user is in a call and presses

Re: [asterisk-users] Joining an astablished call

In the telephony world that is known as barge-in and is a programmable option granting that right to specific extension(s) in systems that normally have automatic privacy. Not all electronic key and hybrid systems have automatic privacy, though most do. John Novack neo haux wrote: Hi, I

Re: [asterisk-users] Joining an astablished call

The best way I have found to do this is to use ChanSpy/ExtenSpy and then use the wisper/barge modes. Jacob From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John Novack Sent: Monday, May 06, 2013 9:46 AM To: Asterisk Users Mailing

[asterisk-users] Installing on an OpenVZ instance

Hello All; I'm attempting to build the dahdi on an OpenVZ instance: Linux serverx 2.6.18-274.7.1.el5.028stab095.1 #1 SMP Mon Oct 24 20:49:24 MSD 2011 x86_64 x86_64 x86_64 GNU/Linux Now, the kernel says that I have the proper one installed, as you can see from above. However, when I run the

Re: [asterisk-users] Installing on an OpenVZ instance

2013-05-06 20:48, James Wystead skrev: Hello All; I'm attempting to build the dahdi on an OpenVZ instance: Linux serverx 2.6.18-274.7.1.el5.028stab095.1 #1 SMP Mon Oct 24 20:49:24 MSD 2011 x86_64 x86_64 x86_64 GNU/Linux Now, the kernel says that I have the proper one installed, as you

[asterisk-users] What is bootstrap.sh for ? Possible bug in 11.3.0 ?

Hi, Before trying to script res-memcached installation (see res_memcachedhttps://github.com/drivefast/asterisk-res_memcached), I banged into this on a fresh 11.3.0 setup: 1. When run for the first time bootstrap.sh displays a non-blocking error. # sh -x bootstrap.sh + uname -sr + MY_AC_VER= +

Re: [asterisk-users] What is bootstrap.sh for ? Possible bug in 11.3.0 ?

On 05/06/2013 05:54 PM, Olivier wrote: Hi, Before trying to script res-memcached installation (see res_memcached https://github.com/drivefast/asterisk-res_memcached), I banged into this on a fresh 11.3.0 setup: snip My questions are: 1. What is the purpose of bootstrap.sh ? 2. Is

[asterisk-users] chan_alsa and confbridge

OK, somebody may have a much better way of doing what I'm attempting. If so, I'm open to suggestions. I am trying to configure confbridge to create a conference room with an audio stream coming from my sound card. The idea is for a group of people to be able to call in and listen to someone

[asterisk-users] НА: asterisk-users Digest, Vol 105, Issue 40

help -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To

[asterisk-users] (no subject)

unsubscribe -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing

[asterisk-users] Get Channel Variables in AMI Event NewExten

Hi, I'm stucked in situation, and look for a work around if possible in Asterisk. I have a dialplan, [default] exten = 111222,n,Set(fu_callerid=141688xyxzz) exten = _X.,n,NoOp(Callerid ${fu_callerid}) exten = _X.,n,wait(2) exten = _X.,n,Answer()   When,  Answer Application is called AMI Event