search for: convergedgroup

Do you know if this can be achieved with the standard sip stack in asterisk? Kind Regards Andrew Colin Converged Telecoms (Pty) Ltd. Licensed Telecoms Operator : (0258/IECS/JAN/09) (0258/IECNS/JAN/09) Switchboard: +27 (0)10?591 4600 Email:? andrew at convergedgroup.net Web: ?http://www.convergedgroup.net 75 Witkoppen Road, Northriding, Johannesburg, 2169 P O Box 7246, Weltevredenpark, 1715 This communication is confidential and intended solely for the addressee(s). Any unauthorized review, use, disclosure or distribution is prohibited. If you believe this mes...

...Outbound calls via a trunk work fine with g729 Kind Regards Andrew Colin Converged Data (Pty) Ltd. Licensed Telecoms Operator : (0258/IECS/JAN/09) (0258/IECNS/JAN/09) Direct: +27 (0)10 591 4607 Mobile: +27 (0)82 310 3007 Switchboard: +27 (0)10 591 4600 Email: <mailto:andrew at convergedgroup.net> andrew at convergedgroup.net Web: http://www.convergedgroup.net 75 Witkoppen Road, Northriding, Johannesburg, 2169 P O Box 7246, Weltevredenpark, 1715 This communication is confidential and intended solely for the addressee(s). Any unauthorized review, use, disclosure or distribution is pr...

Hi Guys I am trying to write a macro for a call return so for example Anyone in the company transfers a call to another extension and it is not answered etc it must return to the person who did the transfer I have got it working but if the call originates externally for example someone calls in to the switchboard and they transfer it then it tries to return to the outside caller. As

Hi Joshua If i put the default_user option per endpoint would it work?? So what exactly does the contact_user option do? I know that in freeswitch there is the option extension-in-contact.We ?basically need to achieve the same functionality? Thanks<div> </div><div> </div><!-- originalMessage --><div>-------- Original message --------</div><div>From:

Hi Guys, We are getting a strange issue on certain polycom phones, sometimes when a call comes in it just "flashes" to show there is a call but does not play any sound. This problem is very intermittent and happens to maybe 2 out of 10 calls. Has any else experienced this before? -------------- next part -------------- An HTML attachment was scrubbed... URL:

...X 300 and 400 phones running UCS 5.2.x. We finally got Polycom to issue a hotfix firmware version. I'll be happy to share it with you offlist, just email me. Officially Polycom will fix the issue in 5.3 in a few months.. Thanks David On Mon, Mar 9, 2015 at 9:34 AM, Andrew Colin <andrew at convergedgroup.net> wrote: > Hi Guys, > > > > We are getting a strange issue on certain polycom phones, sometimes when a > call comes in it just ?flashes? to show there is a call but does not play > any sound. > > This problem is very intermittent and happens to maybe 2 out of 10 ca...

Hi Guys Is it possible to register Kamallio directly to our SIP provider then load balance the RTP to 2 asterisk servers? We cant do the registration from the asterisk boxes as we want to do it directly from Kamallio. Is this possible? -------------- next part -------------- An HTML attachment was scrubbed... URL:

4 Port PRI sangoma a104 From: jg [mailto:webaccounts173 at jgoettgens.de] Sent: Wednesday, March 18, 2015 2:09 PM To: Andrew Colin; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] 4 Port PRI I have a 4 port PRI card that I need to setup each port in their own group. In chan_dahdi.conf I have the following which works for one port How do I add the rest

...it will go thru sometimes. Sent from Samsung Mobile <div>-------- Original message --------</div><div>From: Kevin Larsen <kevin.larsen at pioneerballoon.com> </div><div>Date:16/02/2015 17:11 (GMT+02:00) </div><div>To: Andrew Colin <andrew at convergedgroup.net>,Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com> </div><div>Subject: Re: [asterisk-users] BlindXfer Sensitivity </div><div> </div>> Hi Guys > > We have a client running on a polycom vvx400 IP phone...

Hi I'm using asterisk 1.8 Does anyone know how to use the queue reload command. The built in help doesn't really help. queue reload {parameters|membe Reload queues, members, queue rules, or parameters Regards Ish -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: ish at pack-net.co.uk w: http://www.pack-net.co.uk

This was always possible in the past, however does not work in the current release. I believe this is a bug. To: asterisk-users at lists.digium.com From: cervajs at fpf.slu.cz Date: Fri, 9 Oct 2015 10:04:47 +0200 Subject: Re: [asterisk-users] Storing HANGUPCAUSE in CDR search in archives save the records to another table like cdr_extended Dne

Hi queue reload(queue name) or queue reload all for example queue reload reception From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Ishfaq Malik Sent: Thursday, January 8, 2015 2:10 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] queue reload command Hi I'm using

Hi Guys We have a client running on a polycom vvx400 IP phone on our asterisk 1.8.18 system The issue we have is the switchboard lady uses ## to transfer calls but sometimes it just does not work and just plays the DTMF tone to the calling party. Is there any way to adjust the sensitivity of the blindxfer feature? The polycom Transfer button is useless as there is a big delay

Hi Guys We have a strange issue whereby one phone has delayed rtp So what happens is when the lady answers the phone for the 1st 1 second they can not hear her and then everything is fine I am running asterisk 1.8.28.0 Has anyone seen this before? -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi Guys, I have recently moved my database servers to a different database cluster that runs on haproxy. Every minute or so I get the following error in asterisk MySQL RealTime: Ping failed (2006). Trying an explicit reconnect The strange thing is if I do realtime mysql status It shows as connected just the timer resets. Any idea why this is occurring? -------------- next

Hi Guys I have a 4 port PRI card that I need to setup each port in their own group. In chan_dahdi.conf I have the following which works for one port How do I add the rest of the ports in their own groups so that I can have different signaling on each? [channels] language=en switchtype=euroisdn pridialplan=unknown resetinterval=600 echocancel=yes echotraining=yes

Hi Guys We have a strange a strange issue at a client they have 3 panels on their phone and every so often the panels reboot themselves yet the phone stays on. We decided to replace the T26 for a T28 to see if it fixes the issue and still have the exact same issue. Has anyone seen this before? -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi Guys We are using the wizard to configure our pjsip trunk(see below) How do we get this setting to work contact_user=username We want to change the contact field in the sip invite to display the username of the trunk [trunk_defaults](!) type = wizard transport = transport-udp endpoint/allow_subscribe = no endpoint/allow = !all,g729 aor/qualify_frequency = 30