[asterisk-users] Time Based Voicemail Messages
Hi, I want to change my voicemail message based on the time of day. I would like a message that says Sorry the office is now closed. after a certain time, and says Sorry I am unavailable / Busy / etc before. I have come up with two ways of doing it: 1. A cron job to replace the files (messy) 2. Using different mailboxes at the different times (this means I have 2 mailboxes to check). Is there a way that the voicemail could be enhanced by adding a feature like this? With thanks, Tim ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Time Based Voicemail Messages
Hi, I want to change my voicemail message based on the time of day. I would like a message that says Sorry the office is now closed. after a certain time, and says Sorry I am unavailable / Busy / etc before. I have come up with two ways of doing it: 1. A cron job to replace the files (messy) 2. Using different mailboxes at the different times (this means I have 2 mailboxes to check). Is there a way that the voicemail could be enhanced by adding a feature like this? With thanks, Tim ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Time Based Voicemail Messages
Isn't that covered in point 2? Admittedly, I did not consider using Playback rather than voicemail to play the message. But you didn't point that out anyway. a lil bit of googling wud have answered you Tim. Put in some effort next time anyway, for now : http://www.voip-info.org/wiki/index.php?page=Asterisk+tips+openhours Wildheart wrote: Hi, I want to change my voicemail message based on the time of day. I would like a message that says Sorry the office is now closed. after a certain time, and says Sorry I am unavailable / Busy / etc before. I have come up with two ways of doing it: 1. A cron job to replace the files (messy) 2. Using different mailboxes at the different times (this means I have 2 mailboxes to check). Is there a way that the voicemail could be enhanced by adding a feature like this? With thanks, Tim ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Macro variables and redirects
Hi, I have a dialplan that works like this: ; Arg 1 is the phone, Arg2 is the timeout (optional), Arg 3 is the voicemailbox(optional) exten = 20,1,Macro(dialexten,SIP/1234,15,1234) exten = 21,1,Macro(dialexten,SIP/1235,15) ; Arg 1 is phones, Arg 2 is timeout, Arg 3 is voicemail exten = 30,1,Macro(huntgroup,SIP/1234SIP/1235,25,1234) If I call extension 30, answer it, then redirect to extension 21 via an attended transfer, and no one answers the voicemail will time out to mailbox 1234, when it should not (the macro makes it play busy instead). I can just set a value and test for that (like off), but should the macro arguments for the huntgroup macro be remembered in the dialexten macro like this? The redirect calls the second macro correctly, but if you NoOp the arguments in the macro, you can see that they are inherited from the previous macro) I am using 1.2.10. With thanks, Tim ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Dynamic Codec Selection
Hi, The PSTN connection is via a zaptel card, rather than a sip peer. With thanks, Tim On 2006-10-24 06:44:01 -0700, Wildheart [EMAIL PROTECTED] said: Hi, Does anyone know a what to use a different codec for calls which a re handset to handset (eg, G711) then when we have calls to the out side world (via an asterisk server) to use a different codec(eg, G729)? snip I responded: yes, this is simple, just make it so the extensions allow both g729 and ulaw, and set your outside world is g729. On 2006-10-25 03:31:39 -0700, Wildheart [EMAIL PROTECTED] said: Hi Marty, By the outside world, I mean the PSTN connection. I am still intereste d in how you would set this up. Can you paste in a sample config? One internal phone from SIP.conf: ; ; SIP entry for users test rig [2004] type=friend secret=footest dtmfmode=inband ; my stupid PSTN gateway doesn't like rfc2833 auth=md5 host=dynamic nat=yes canreinvite=no disallow=all allow=ulaw allow=g729 context=autocontext callerid=Alton Wireless Phone 2004 Another internal extension ; IAX entry for user karma [3000] type=friend secret=testfoo auth=md5 host=dynamic disallow=all allow=ulaw allow=g729 context=karma callerid=Karma206500 Ok, now these two extensions when one calls the other should use uLaw. Now here is my extension for my PSTN gateway: ; ; SIP entry for user (FXO) [2003] type=friend secret=testPSTN dtmfmode=inband auth=md5 host=dynamic nat=yes canreinvite=no disallow=all allow=g729 context=autocontext callerid=Alton Qwest Line2065551183 Depending how your PSTN is setup the last bit could be quite different, but the premise is the same. Since the PSTN only allows g729, this will force other connections to that also. Of course you need to be sure your devices support this, or else you will need to buy licenses for G729 to transcode, which is also a significant hit for CPU. Further more, this only makes sense to do if your PSTN calls are being terminated by someone OFF your local network. If your PSTN calls (like mine) are being routed to a local gateway, then using ulaw should be ok also (it's your network, make it work!). Marty ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Dynamic Codec Selection
Hi Marty, By the outside world, I mean the PSTN connection. I am still interested in how you would set this up. Can you paste in a sample config? With thanks, Tim On 2006-10-24 06:44:01 -0700, Wildheart [EMAIL PROTECTED] said: Hi, Does anyone know a what to use a different codec for calls which are handset to handset (eg, G711) then when we have calls to the out side world (via an asterisk server) to use a different codec(eg, G729)? The idea is to reduce the bandwidth to the server for the majority of calls, but get good quality on internal calls. With thanks, yes, this is simple, just make it so the extensions allow both g729 and ulaw, and set your outside world is g729. Marty ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dynamic Codec Selection
Hi, Does anyone know a what to use a different codec for calls which are handset to handset (eg, G711) then when we have calls to the out side world (via an asterisk server) to use a different codec(eg, G729)? The idea is to reduce the bandwidth to the server for the majority of calls, but get good quality on internal calls. With thanks, Tim ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MixMonitor Problems
Hi, I am running * 1.2.9.1 on a server recording calls via MixMonitor. I have recorded one call which according to the cdr logs was 40 minutes, but the recording seems to stop after 22. I know this problem was fixed ages ago, but has anyone else noticed this? Any idea what could be causing it? With thanks, Wildheart ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Echocancelwhenbridged
Hi, Can someone tell me what are the valid parameters for the option echocancelwhenbridged? Is it just yes or no, or does it support 128 as well? Also is thier any differnce with using echocancelwhenbridged=128 as opposed to echocancelwhenbridged=yes (assuming that 128 is a valid option). With thanks, Tim ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Caller ID Matching in extensions.conf
Hi Doug,
Shouldn't you really be using ${CALLERID(number)}?
Also, if the channel you are using to get the caller ID from is analog (FXO
or FXS), I believe you may have to answer the channel, then wait 1 sec to
get the correct caller id info.
Tim
- Original Message -
From: Douglas Garstang [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Friday, June 23, 2006 8:28 PM
Subject: [Asterisk-Users] Caller ID Matching in extensions.conf
I'm running 1.2.9.1, and I can't get caller id dialplan matching to work.
When calling from 9220370 to 1234, the following does not match.
exten = 9220370/1234,1,NoOp(${CALLERIDNUM})
exten = 9220370/1234,2,Answer
exten = 9220370/1234,3,Playback(tt-weasels)
However, when calling from 9220370 to 1234, this DOES match.
exten = 1234,1,NoOp(${CALLERIDNUM})
exten = 1234,2,Answer
exten = 1234,3,Playback(tt-weasels)
You can also see from the console output that the caller id IS 9220370.
-- Executing NoOp(SIP/9220370-7a11, 9220370) in new stack
-- Executing Answer(SIP/9220370-7a11, ) in new stack
-- Executing Playback(SIP/9220370-7a11, tt-weasels) in new stack
-- Playing 'tt-weasels' (language 'en')
What am I missing here?
Oh, this also doesn't match EVER... so I am wondering if there's a problem
with dialplan caller id matching in 1.2.9.1?
exten = _X./1234,1,NoOp(${CALLERIDNUM})
exten = _X./1234,2,Answer
exten = _x./1234,3,Playback(tt-weasels)
Doug.
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Re: [Asterisk-Users] Caller ID Matching in extensions.conf
D'oh... Might have just answered the wrong question here... Also, if the channel you are using to get the caller ID from is analog (FXO or FXS), I believe you may have to answer the channel, then wait 1 sec to get the correct caller id info. Tim ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Conferencing with multiple servers
Hi, I am trying to join 2 asterisk servers together using a sip channel. This is so, if a user joins a conference on box A and another user joins a conference on box B, providing they are in the same conference room, the two conferences are joined via the sip channel. We only want to join the conferences together if they have users in them and we don't want to point all the conferences to one server as we would like to try to balance the load a bit. Any ideas on how to impliment this? With thanks, Tim ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] manager DBDel action
Hi, Have a look at this ticket: http://bugs.digium.com/view.php?id=6874 It contains the patch to add dbdel to your implimetation, but the command is not being added to the core of asterisk. Tim Hi list, is there a possibility to delete a key from the astdb through the manager interface? I managed to put and to get a key but I do not know how to delete an entry. The problem is that I want to use the manager interface because I can communicate remotely with my * this way. TIA, Christophorus ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
