similar to: Call Limits

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 =>

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 everyone, Does anyone have the documentation for OutCall? http://code.google.com/p/outcall/ The link isn't working. 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

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, 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

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

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, 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

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>

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

A little more thought into helping troubleshooting maybe take an ls -l and a properties screen shot in the middle of editing a file. It could be some setting in notepad++ that is interfering with itself. On Thu, Apr 7, 2016 at 8:51 AM, Jeff Sadowski <jeff.sadowski at gmail.com> wrote: > Can you post ls -l of a file on your samba shared directory.(before > opening it with notepad++)

If your file is unchanged and the modified time is unchange in both ls -l and the Windows file browser and notepad++ saw a change then the problem is with notepad++ not in samba or linux. On Thu, Apr 7, 2016 at 10:26 AM, RITTER, Philippe < philippe.ritter at caisse-des-medecins.ch> wrote: > Thank you for your mail. > > Just one mor information, I asked help also on notepad++ :

I want to set up a conference call to be recorded automatically, so I'd like the recording to start when the second caller joins the conference (one caller already there). The recording would continue until the last user hangs up. How can you determine how many are already in the conference bridge? [conferences] exten => 66,1,Answer exten => 66,n,Wait(1) exten =>

Hi, is there a way to limit an account to do simultaneous calls in sip and iax? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070713/841cbb5f/attachment.htm

I'm running Asterisk without FreePBX or any of the other managers. I'm trying to figure out how to set the maximum number of channels allowed on a single line? I'd just rather not have Asterisk try the line when I know I'll recieve a CONGESTION back from the SIP phone provider (ViaTalk in this case). Is there a configuration option I can't find that sets the maximum number

Hi to all, I would like to limit the numbers of inbound h323 connections for different extensions, for instance, I've the following rules in my dialplan: exten => 123,1,DIAL(H323/1100) exten => 234,1,DIAL(H323/2200) and I would like to limit to 5 the number of h323 connections for exten 123 e to 2 those for 234. The reason to limit the number of connectios is that these h323

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

Is there a way to limit simultaneous calls. I like to limit simultaneous outgoing calls as more than few simulataneous calls are charged by my voip providers. However, I do not want to have any such restriction for internal calls. Thanks Jim