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Rebooting pfsense solved the problem, but it is no good feeling, if you donĀ“t know if your phone will work :-(( I am still interrested in ideas and suggestions of the cause of my problem! Regards, Karl


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As of 2014, this needs to use Skype Connect, the other methods used when this was originally asked are no longer available


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Turns out that there is an undocumented event called QueueMember. Within this event you'll find the paused attribute. In my case, this event was used as the total queue member counter, at which was adding all queue members to the non-paused count on the server. Here is my source: http://forums.digium.com/viewtopic.php?f=1&t=83884


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The "ASTERISK_PROMPT" is an environment variable and should be set prior to starting the CLI for it to take affect. i.e. export ASTERISK_PROMPT="%C31%H*CLI%#" asterisk -r As for your specific version, (Asterisk 11 branch) The reasons for not displaying colors is as follows: The command-line options to connect to the server includes the "no-color" flag. ...


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A GSM FXO is no different to an analogue FXO in terms of operation, the difference is only in the medium for which it connects. I have one on my network which I use to connect my Analouge BT phone line into my Asterisk PBX service, so that when inbound calls come into my fixed UK land line, they get picked up by asterisk and handled appropriately. In my ...


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Main issue with t.38 are following 1) by default asterisk compliled without spandsp support, so no t.38 2) even if you comiled that, you t38udpl line have be EXACT same as provider expect. Any single param mistake can result not working t.38 3) on outbound call t.38 have be initiated by CALLED side. So if your provider NOT detect fax or NOT ask asterisk ...


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That mean you have no port forwarding on external router, SIPALG enabled on router or not setuped NAT correctly. For more info see sip debug and study this article http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions


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1) can be done by changing "Asterisk Dial Options" in "Advanced features" tab. You need add S(1200) S(x): Hang up the call seconds after the called party has answered the call. 2) no, you can't in default elastic install. However you can request expert do module for such task( need add into sip_additional.conf variables and use that variables ...



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