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...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/web/digisign/digisign.asp?login=I23E7F508C6B61A91700343&lang=en&surpre=PreSurvey> This email and any files transmitted with it are confidential and intended so...

...t 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/attachments/20110317/f35c9343/attachment-0001.htm>

Hi, Does anyone know where this is suddenly coming from? -- Remote UNIX connection -- Remote UNIX connection disconnected -- Remote UNIX connection -- Remote UNIX connection disconnected -- Remote UNIX connection -- Remote UNIX connection disconnected Thanks Dan p.s. sorry about the last post. hit the mouse by mistake and it sent the email. -------------- next part

Hi, We have SPA921 handsets which apparently support Paging, however i can't find any information on configuring Asterisk to make a page call. Does anyone have any information on Paging? Many thanks Dan Journo -------------- next part -------------- An HTML attachment was scrubbed... URL:

...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/web/digisign/digisign.asp?login=I23E7F508C6B61A91700343&lang=en&surpre=PreSurvey> This email and any files transmitted with it are confidential and intended so...

...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/web/digisign/digisign.asp?login=I23E7F508C6B61A91700343&lang=en&surpre=PreSurvey> This email and any files transmitted with it are confidential and intended so...

...>> 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: > <31C6BA8C3525D840B022617ACB...

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

Hi, Im looking at using MixMonitor to record calls and I know that I need to set the filename first. However, with the number of calls coming in, hard coding the filename isnt an option. So I need to do something like this:- MixMonitor(RANDOMNUMBER.wav) But can't find a way to generate a random number. I thought that maybe I could use a unique variable that already exists for the current

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-Encoding: quoted-printable <html xmlns:v=3D"urn:schemas-microsoft-com...

...ent 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

Hi, I think I already know the answer to this question, but is there any way to do the following using realtime? Or do I have to create a full dialplan for each client without using includes? [client1_phones] include => client1_internal include => client1_outgoing_calls include => test_calls include => parkedcalls [client2_phones] include => client2_internal include =>

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

Hi, I've got a client with two ADSL connections for redundancy. Is it possible to set up asterisk to connect to one SIP provider using both adsl connections and load balance between the two connections? Or to use one connection as the main one, and automatically fail over if the first connection drops? Or does this kind of thing need a serious network switch? Thanks Dan --------------

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 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

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

Hi, I was wondering if anyone is successfully running Asterisk in a cloud environment. If you could state which cloud you are using, I'd appreciate it. Many thanks Dan Journo -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091014/076ff188/attachment.htm

Hi, Which DTMF mode do people mostly use? I've tried SIP INFO and RFC2833 but although Asterisk recognises the tones (for feature usage), the tones arent repeated to the end user. So if I call a company that has a menu system, I can't use the menu. Thanks Dan -------------- next part -------------- An HTML attachment was scrubbed... URL: