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...orking. Thanks Dan ________________________________ IT Maintenance Contract Clients can now access our Instant Chat Service to receive immediate remote IT support. Click the chat link below for support. For more information on receiving IT support from ?150 per month, please contact Kesher Communications. ________________________________ Dan Journo IT Business Consultant Kesher Communications Ltd Tel: 07957 233 599 Web: http://www.KesherCommunications.com <http://www.keshercommunications.com/> Live Chat/Instant Support: Click Here <http://eu.ntrsupport.com/inquiero...

...alls. Many thanks Dan ________________________________ IT Maintenance Contract Clients can now access our Instant Chat Service to receive immediate remote IT support. Click the chat link below for support. For more information on receiving IT support from ?150 per month, please contact Kesher Communications. ________________________________ Dan Journo IT Business Consultant Kesher Communications Ltd Tel: 07957 233 599 Web: http://www.KesherCommunications.com <http://www.keshercommunications.com/> Live Chat/Instant Support: Click Here <http://eu.ntrsupport.com/inquiero...

...? Thanks Dan ________________________________ IT Maintenance Contract Clients can now access our Instant Chat Service to receive immediate remote IT support. Click the chat link below for support. For more information on receiving IT support from ?150 per month, please contact Kesher Communications. ________________________________ Dan Journo IT Business Consultant Kesher Communications Ltd Tel: 07957 233 599 Web: http://www.KesherCommunications.com <http://www.keshercommunications.com/> Live Chat/Instant Support: Click Here <http://eu.ntrsupport.com/inquiero...

...26 I am using the "s" option with ChanIsAvail because if I run the page, it interrupts the current call. I need to give the caller the busy signal when a minimum of 1 channel is in use. Can anyone suggest how to test whether the user is on the phone? Many thanks Dan Journo Kesher Communications Ltd -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091103/d93e6123/attachment.htm

...ueue members specifically log out of a queue. I've looked at autopause, but we need it to automatically un-pause once it comes back online. Any idea how I can do this? Preferably without using the AMI or AGI scripts, but if that's the only way, then i'll have to use that. Thanks Dan Kesher Communications (UK) Business Phone Systems<http://www.keshercommunications.com/> | Hosted PBX<http://www.keshercommunications.com/hostedpbx.html> -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachm...

...speak to each other without having to wait for one of the 2 other calls to end. I thought that maybe one way would be to duplicate the outbound sip settings and label them "outbound_client_1" and then use call-limit within that. Has anyone got any experience of this? Thanks Dan Journo Kesher Communications Ltd -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091206/7f39cef7/attachment.htm

a phone system, and plug it into a SIP Adapter like the PAP2T. Never done it myself, so I can't recommend a suitable intercom. Hopefully s= omeone else can. Dan Journo Kesher Communications (UK) Business Phone Systems<http://www.keshercommunications.com/> | Hosted PBX<h= ttp://www.keshercommunications.com/hostedpbx.html> --_000_31C6BA8C3525D840B022617ACBB7BC0383532DB40BVMBX123ihoste_ Content-Type: text/html; charset="us-ascii" Content-Transfer-...

Hi I'm using asterisk 1.8.3.2 (with a couple of patches) I have the following scenario... SIP call comes in and gets answered by extension A (MixMonitor is executed as part of this inbound dial plan of the number being called) Extension A puts call on hold and calls extension B Extension A then does an attended transfer of incoming call to extension B I'm finding that the recording

Hello Everyone; I am looking for a solution to record calls that come into our Asterisk server. I am hoping for something that is easy to use - however, if I have to modify it to make it easier to use, I do not mind. Does anyone know of any opensource or otherwise solutions out there that I can try out? Thanks much. Glen

...>> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users > > > > ------------------------------ > > Message: 6 > Date: Sat, 9 Apr 2011 20:45:58 -0400 > From: Dan Journo <dan at keshercommunications.com> > Subject: Re: [asterisk-users] Call Recording using MixMonitor - close, > but would like some more words of wisdom. > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users at lists.digium.com> > Message-ID: > <31C6BA8C3525...

...and N510P) and Asterisk? A client has them, and whenever they try a blind transfer, something goes wrong. Agent 1 starts and completes the blind transfer. Agent 2 answers the transferring call. Agent 2 hears asterisk music on hold, but the caller can hear the agent. Any ideas? Thanks Dan Journo Kesher Communications (UK) www.keshercommunications.com<http://www.keshercommunications.com> -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20140212/dbd8f6f0/attachment.html>

Hi, could anybody suggest a usable doorphone and magnetic door opener "hardphone" system for me, please? Of course should be connectable to asterisk. I am in the EU, should be available here. thank you, Csaba

Dan et al; This looks like a perfect solution. However, I have one issue. If I initiate the macro manually (put it in the proper context/dialplan) it works. I see the *.wav file being created and growing in the /var/spool/asterisk/monitor directory. If I try to implement it adding the MixMonApp => *1,self/both,Macro,mixmon to the [applicationmap] in features.conf, I cannot get it to

Hello, My host just had a faulty power supply and therefore, my Asterisk server was down for 7 hours. It was a Sunday so no one was making calls, however if it happened during the week, I'd have problems. I was trying to find a whitepaper or advice on how to set up two Asterisk servers to provide some redundancy. I've been googling "asterisk redundancy" but all I've found

Hi, Would you recommend some standalone SIP phones that work well with Asterisk? Personal experience preferred. Thanks, -- Raj

...il", $options); >> >> now the question is how to I get the VoiceMailMain to not ask for >> "Mailbox" >> and already know which mailbox and just prompt for "Password" >> >> >> On Thu, Apr 7, 2011 at 6:44 PM, Dan Journo <<dan at keshercommunications.com> >> dan at keshercommunications.com> wrote: >> >>> > Unfortunately, that solution will not work for me... The user must be >>> able to hit * during the greeting of any voicemail and be prompted for >>> the >>> "Password...