similar to: OutCALL

Hello, I was wondering if anyone has any insights on the best way to automatically monitor an asterisk box to check it is constantly available and processing calls. 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

Hi, Does anyone have an up to date guide for setting up fax 2 email with asterisk? 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

Hi all, I am having a problem with ChanIsAvail. It always returns the same result, regardless of whether an extension is available or not. It always returns 0 Unknown Status. This is my dialplan. exten => _2XX,1,ChanIsAvail(SIP/winsor_${EXTEN}|s) exten => _2XX,2,Verbose(0, ${AVAILSTATUS}) exten => _2XX,3,GoToIf($[${AVAILSTATUS} = "1"]?4:5) exten =>

Hi, I'm trying to work out an issue with call queues. I need the calls that are in a queue to be kicked out if all members are unavailable (for example if all SIP members are having network problems). I tried leavewhenempty = yes but that only seems works when all queue members specifically log out of a queue. I've looked at autopause, but we need it to automatically un-pause once it

Bicom Systems releases outCALL, an Asterisk open source Outlook integration LONDON, UK (11th April 2007) - Bicom Systems announced today it has released outCALL, an open source desktop application allowing integration Microsoft Outlook. OutCALL allows users an easy way for placing and receiving phone calls integrated with users Outlook contacts. "The open source PBX market needed

Hi OutCALL 1.40 is released. It is available in two flavours: - Without extension authentication - With extension authentication Changelog: OutCALL 1.40 (2007-06-29): - Multi-language support (French-Canada is included in the setup, while the English PO file is distributed with OutCALL setup which can be translated and added into OutCALL in run-time) Please use http://www.poedit.net/ for

Hi Everyone, I am using OutCall 1.6 (latest) with Asterisk 1.6 and Windows Vista. I can originate calls see the program login nicely but when a call comes in it only shows the Name portion of the CLID and not the number hence it pulls up a new contact on Outlook. The new contact only show name and last name and no CLID Number again. So, this repeats every-time I call even if I manually enter a

Has anyone implemented an outcall notification when there is a voice message waiting? I would like to have the system notify me of awaiting voice messages by a telephone call rather than an email notification. I would imagine that a call could be dumped into the asterisk spool directory, but I'm not sure how I would monitor for messages waiting. Has anyone implemented such a feature for

Hi actually, for out call, i use : exten => _0XXXX.,1,Dial(SIP/out-l1/${EXTEN:1},50,rt) exten => _0XXXX.,2,Dial(SIP/out-l2/${EXTEN:1},50,rt) exten => _0XXXX.,3,Dial(SIP/out-l3/${EXTEN:1},50,rt) exten => _0XXXX.,4,Hangup can you say me with this config, if the first user call and use out-l1 the second user use automatiquely out-l2 (and out-l3 when l1 and l2 are used) ? if i want

Hi, as far as I know my problem is not a bug but wanted behaviour. Let's assume the following dialplan: exten => 123,1,Answer exten => 123,n,Dial(DAHDI/g0/00492112233,20,g) [...] exten => 123,n,Hangup I do an dial-in with my SIP-Client (or phone). The Dial-Application starts the outdial and I get connected with my partner. After hangup, only 1 CDR is generated. But I(!) need 2

Hello, I'm trying to figure out how to limit the number of concurrent calls a client can make. I have a client that has 6 SIP accounts. One for each SIP phone. I want to limit it so that they can only make 2 outgoing calls at a time so that I can bill them "per channel" rather than "per extension". A separate (but not so important) issue is that I want them to be able to

Resent to the list since I didn't think you would mind. Kevin wrote: > Eric, > > I have been using your vm outcall script for some time and it has worked > well. Thanks for your efforts. > > I am trying to re-install and I can't seem to get a call file generated. > I have set up postfix and in the log it appears that it pipes the > message to the vmoutcall

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>

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

please send me the ways to change asterisk port from 5060 to 5061 i need to configure it because we are already using 5060 port in router then we cant use it again we have to configure other sip server so please suggest me a way.......................... On 4/10/11, asterisk-users-request at lists.digium.com <asterisk-users-request at lists.digium.com> wrote: > Send asterisk-users mailing

Hi, Is anyone aware of an issue with Gigaset DECT handsets (R630H 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)

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