Try:
; Dial an extension
exten = _1.,1,Dial(SIP/${EXTEN:1})
That would match anything entered starting with a 1 and dial it without the
first digit.
Jeremy Koski wrote:
Here's what I'm trying to accomplish:
Press 1 to transfer to extension
Press 2 for Directory
Press 0 for Operator
Asterisk only needs a cLock to play MOH afaik, on 2.6.x kernels you
don't need any timing help, on 2.4.x you can use ztdummy on the USB drivers
Damon Estep wrote:
Sounds like you lost timing, either because a zaptel device driver did
not load or ztdummy did not load if you have no zap
Here, have a couple of my s ;)
Steve Langstaff wrote:
Here, have an 'l' - I've go a couple spare on my keyboard :)
I guess it needs a clock to play sounds...
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Damon Estep
Sent: 31
That should be controllable by a weight, for example 2 peers:
A -- G729, G711
B -- G711, G729
What's currently happening is that * starts transcoding between the two
(g729 for A and G711 for B), what i would like is to apply a weight to
peer A so that the codec of choise at both sides becomes
If you're using Cisco Switches:
Logon to the switch and go to config mode
int fa0/1
switchport access voice vlan untagged
sometheing in that direction configures the CDP to set the phone to
untagged frames.
Julian Lyndon-Smith wrote:
How ? Where ? I've been wanting to do this for ages, and
Rich Adamson wrote:
I though that Asterisk would transcode between codecs! All my SIP devices
support G729a
711a/u but my ISDN PRI/BRI lines use 711a/u. Asterisk is quite happy to accept
a call from a SIP
device using G729a and then complains that it can't translate into G711 to go
What happens if the rate changes mid call?
IE, call starts @ 18.30 and lasts till 19.15
Rate changes @1900 to off-peak.
Darren Wiebe wrote:
Partially. I have not finished the script that will limit the calls
depending on the money available.
Darren Wiebe
[EMAIL PROTECTED]
VoIP Newbie
I'm having some troubles with my * machine, when i place a call on hold
the callee doesn't hear any MOH and the call is dropped because of lack
of RTP.
I also don't see * starting MOH on the SIP channel the callee is on (moh
class is defined, there are MP3 files and mpg123 is active).
I'm using *
That would be nice, SER does have the possibility to answer an OPTIONS
correctly, but * indeed answers with a 404, i'm now using sipsak to
register a test user, that also works.
Andres wrote:
Erik Versaevel wrote:
Hello all,
Can asterisk itself respond to a SIP OPTIONS packet? i wan't to check
Hello all,
Can asterisk itself respond to a SIP OPTIONS packet? i wan't to check if
asterisk is still alive by using sipsak (because of nagois mon)
Kind regards,
E. Versaevel
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Mark Elkins wrote:
On Tue, 2005-01-04 at 15:45 +0200, Mark Elkins wrote:
On Tue, 2005-01-04 at 15:20 +0200, Mark Elkins wrote:
I've got asterisk able to make and receive calls via the Internet via
E164 lookups. If I get such a call - I'd like to display the original
Playing with
Sorry for my hars anwer :)
You are able to call into asterisk (moh for example) using G729?
Roy Sigurd Karlsbakk wrote:
I did read the docs and I did purchase the G.729 license.
sipgw1*CLI show g729
0/0 encoders/decoders of 25 licensed channels are currently in use
roy
On Dec 28, 2004, at 9:31,
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