Displaying 6 results from an estimated 6 matches for "thtr".
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2005 Jun 29
4
Music oh hold
...gium.com
Oggetto:
Hi, I installed mpg123 v0.59r without error and defined as defaut folder /var/lib/asterisk/mohmp3. When i set a call on hold everythinghs seem ok, but i cannot hear music. I'm using asterisk 1.0.8
*CLI> -- Executing Dial("SIP/2339-4da6", "SIP/2391|60|Thtr") in new stack
-- Called 2391
-- SIP/2391-79a0 is ringing
-- Saved useragent "PA168S" for peer 2319
-- SIP/2391-79a0 answered SIP/2339-4da6
-- Attempting native bridge of SIP/2339-4da6 and SIP/2391-79a0
-- Started music on hold, class 'default', on SIP...
2005 Jun 30
3
R: Music oh hold
...ium.com
Oggetto:
Hi, I installed mpg123 v0.59r without error and defined as defaut folder /var/lib/asterisk/mohmp3. When i set a call on hold everythinghs seem ok, but i cannot hear music. I'm using asterisk 1.0.8
*CLI> -- Executing Dial("SIP/2339-4da6", "SIP/2391|60|Thtr") in new stack
-- Called 2391
-- SIP/2391-79a0 is ringing
-- Saved useragent "PA168S" for peer 2319
-- SIP/2391-79a0 answered SIP/2339-4da6
-- Attempting native bridge of SIP/2339-4da6 and SIP/2391-79a0
-- Started music on hold, class 'default', on SIP...
2005 Jun 29
0
(no subject)
Hi, I installed mpg123 v0.59r without error and defined as defaut folder
/var/lib/asterisk/mohmp3. When i set a call on hold everythinghs seem
ok, but i cannot hear music. I'm using asterisk 1.0.8
*CLI> -- Executing Dial("SIP/2339-4da6", "SIP/2391|60|Thtr") in new
stack
-- Called 2391
-- SIP/2391-79a0 is ringing
-- Saved useragent "PA168S" for peer 2319
-- SIP/2391-79a0 answered SIP/2339-4da6
-- Attempting native bridge of SIP/2339-4da6 and SIP/2391-79a0
-- Started music on hold, class 'default', on SIP...
2003 Aug 19
3
MusicOnHold
Does anybody know why I can NOT hear the MusicOnHold - using SJphone on
another PC in our network (normal playback is not a problem) .
See the * output and the line configured in extension.conf below (also
mp3player does not function)
Any suggestions?
*Asterisk output:*
*CLI> -- Executing WaitMusicOnHold("SIP/jeroen-bf54", "30") in
new stack
--
2005 May 30
2
R: R: AT-320 + supervised transfer
The procedure that will do asterisk is very nice ;) but whe it was available ?
Currently is there any way to emprove the transfer? I tryied the scenario that u suggest me but it doesn't work :| and i don't why.
Here my sip.conf for the phone, can u say me if there is somethingh wrong ?
[2391]
type=friend
username=2391
secret=2391
language=it
host=dynamic
context=intern
dtmfmode=rfc2833
2005 May 30
3
R: AT-320 + supervised transfer
Hi,
Thanks for yuor answer.
The boot time of the phone is very very fast, 10 sec to startup and 2 or 3 second to login to asterisk. I set the NTP server to 255.255.255.255 so it don't try to get time.
I thinked carefully to your scenario and i am going to try it, but i don't known if it could like to my customer
I will try also to use CVS, but i am skeptic to utilize asterisk to