[asterisk-users] Broadsoft - Asterisk interop

2014-02-27 Thread Stelios Koroneos
Greetings to all. I am not sure of this is a user question or a business so apologies in advance if it should be asked in the business list. A client of mine has a UK branch that is served by a provider that uses the Broadsoft solution. I want to create a sip trunk from a remote asterisk pbx to

Re: [asterisk-users] chan_sip.c:3115 __sip_xmit of 0x108d33c0 (len 523) to xxx.xxx.xxx.xxx:0 returned -1: Invalid argument

2014-02-27 Thread Ishfaq Malik
Hi Re raising this issue as it's still affecting me. Where is the asterisk server getting port 0 from? We use ARA and port 0 is neither in the full contact not in the port field of the sip table. Nor is port 0 in the realtime cache for any peer registering from the IP address generating the

[asterisk-users] Asterisk 12 - 100rel (Prack) no 100rel Require in responses

2014-02-27 Thread JULIAN RUSSELL
Hi I am trying out Asterisk 12 between two servers in order to test the 100rel (PRACK) capability. When I set the called server to send Ringing (180) and wait before answering the call it is not sending 100rel Require in the Ringing message (non provisional response) therefore the calling

[asterisk-users] Temporarily placing confbridge participants on hold - two way muting

2014-02-27 Thread Steve Hanselman
Is there a way of temporarily suspending participants in a conference? Say I have 5 users, A,B,C,D,E and I wish A, B and C to have a discussion in the confbridge session that D and E can't hear, is there a way to suspend D and E for a while (whilst they are played music or whatever) and later

Re: [asterisk-users] Asterisk 12 - 100rel (Prack) no 100rel Require in responses

2014-02-27 Thread Joshua Colp
On 14-02-27 06:26 AM, JULIAN RUSSELL wrote: Hi I am trying out Asterisk 12 between two servers in order to test the 100rel (PRACK) capability. When I set the called server to send Ringing (180) and wait before answering the call it is not sending 100rel Require in the Ringing message (non

Re: [asterisk-users] Realtime Call Queues : call members in certain order

2014-02-27 Thread Jonas Kellens
On 13-02-14 17:33, Steven Wheeler wrote: *From:*asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Jonas Kellens *Sent:* Thursday, February 13, 2014 7:12 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re:

Re: [asterisk-users] Temporarily placing confbridge participants on hold - two way muting

2014-02-27 Thread Tech Support
You may want to check out the 3rd party Asterisk module app_konference. You can find it at http://sourceforge.net/projects/appkonference. I have customers using it for the last year or so with very few problems. One customer is routinely running conferences with 80 - 100 users on a Pentium 4

[asterisk-users] G729 Licensing Revisited - I'm Sorry!

2014-02-27 Thread Jayson Devor
Hello Everyone, We are looking to transition our 23 channels from testing/lab into production. During testing we used the free open source g729 license using the instructions found here: http://blog.manhag.org/2010/05/installing-the-free-g729-codec-for-asterisk/ A little more about our setup.

Re: [asterisk-users] G729 Licensing Revisited - I'm Sorry!

2014-02-27 Thread Paul Belanger
On Thu, Feb 27, 2014 at 5:24 PM, Jayson Devor jayson.de...@gmail.com wrote: Hello Everyone, We are looking to transition our 23 channels from testing/lab into production. During testing we used the free open source g729 license using the instructions found here:

Re: [asterisk-users] G729 Licensing Revisited - I'm Sorry!

2014-02-27 Thread Darryl Moore
On Feb 27, 2014 10:02 PM, Paul Belanger paul.belan...@polybeacon.com wrote: No such thing as 'free open source g729 license', if you actually read the site: There is regarding the copyright on the code. The fact it is also patent encumbered is a different issue. DISCLAIMER: You might have