this post is attached to the prevoius post, this is what i have on CLI when i
call from Linksys pap2t to asterisk and then asterisk bridge the call to a sip
provider:
-- Executing [88017736288...@direct:1] Dial(SIP/490115-092bacc8,
SIP/us/88017736288155) in new stack-- Called
I did not really spend too much time on looking at call forwarding and
wonder if someone could help me.
It seems that for setting call forwarding on the Polycom phone itself,
only forward all calls will work. The other call forward function
like forward if no-answer for n rings or forward if
Hi,
The descriptions of Progress() and Proceeding() are really vague.
Progress():
---cut
[Synopsis]
Indicate progress
[Description]
Progress(): This application will request that in-band progress information
be provided to the calling channel.
---cut
On Fri, Feb 13, 2009 at 2:37 PM, John Todd jt...@digium.com wrote:
On Feb 13, 2009, at 11:19 AM, Philipp von Klitzing wrote:
Hi there,
is gizmo the first user of the Digium Skype solution, or do they use a
different approach/product - any clue?
http://www.gizmo5.com/pc/opensky/
At 07:28 AM 2/14/2009, Philipp Kempgen wrote:
But OTOH Indicate progress or Indicate proceeding doesn't
mean anything for the end user.
183 starts early media, 100 does not.
Are there any situations where it makes sense to use either of
these applications from the dialplan?
Philipp Kempgen
Mike wrote:
I've read some sources claiming that Asterisk does not need DAHDI for
timing in 1.6.1. Is this true? Searching the web, all I can find is
pages celebrating the fact but no actual documentation on which version
it was introduced in and how one would go about configuring an
Hi,
We are having a strange issue. If we run asterisk from /etc/inittab
and then connect using asterisk -r, we don't see any logs coming in
CLI. However logs are properly reported to /var/log/asterisk/messages
and system is working fine. Now, if we run from command line (asterisk
-f) and then
joek...@gmail.com schrieb:
Default FreePBX context,
[from-pstn]
The call seems to be going here
[ext-did-catchall]
So?
Philipp Kempgen
--
AMOOCON 2009, May 4-5, Rostock / Germany - http://www.amoocon.de
Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de
AMOOMA
Tilghman Lesher schrieb:
On Friday 13 February 2009 09:55:49 cbbs...@hotmail.com wrote:
I wrote a PERL AGI script that prompts a caller to leave a message using
print RECORD FILE $recordfile wav # 6 BEEP s=3\n;
When the caller is done, they need to press the # key. The message is then
Man, as the CLI says:
SIP/us-092acb78 is ringing (here it gives me a fake ring)
It's the channel SIP/us/something, which is generating ring signalling.
2009/2/14 wassim Darwish wassim...@hotmail.com
this post is attached to the prevoius post, this is what i have on CLI
when i call from
Yes, in fact I usually disable the Call Forward button on phones if possible.
On the Polycoms you could assign a BLF that will show the status.
I usually use a dialplan that will tell the caller thru prompts the
callforward status, like Call Forwarding no answer is currently
enabled to 1234
On
Hi All,
If i buy 20 g729 and install to my asterisk, if 20 calls are already
engaged using g729. would the next call then revert to using the other
codec, in this case ulau and alaw?
thank you
regards,
nhadie
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On Wed, 11 Feb 2009 12:34:12 +0100, Lenz Emilitri wrote:
This is a bit of trickery, but could not resist :)
This will kill a channel that is connected to SIP/201
asterisk -rx soft hangup $(asterisk -rx 'show channels' | grep SIP/201
| awk '{ print $1 '} )
what if there're also channels
Julian Lyndon-Smith ha scritto:
If I have the following in the dialplan
exten = foo,n,Dial(SIP/1234Zap/G1c/55443322)
and SIP/5432 calls this extension,
is it possible to show different callerid numbers to each of the target
numbers ?
The reason I ask is that if the call is from an
On Fri, Feb 13, 2009 at 06:08:45PM +0800, Dinesh Nair wrote:
On Wed, 11 Feb 2009 12:34:12 +0100, Lenz Emilitri wrote:
This is a bit of trickery, but could not resist :)
This will kill a channel that is connected to SIP/201
asterisk -rx soft hangup $(asterisk -rx 'show channels' |
Hi, can anyone help. Bit correction in previous message, there are no
logs in /var/log/asterisk/messages too.
Thanks
On Sat, Feb 14, 2009 at 8:58 PM, Jim Boykin boykin...@gmail.com wrote:
Hi,
We are having a strange issue. If we run asterisk from /etc/inittab
and then connect using asterisk
the problem is, when asterisk boots, it produces logs and are recorded
properly. However, after that there are no logs
On Sun, Feb 15, 2009 at 12:59 PM, Jim Boykin boykin...@gmail.com wrote:
Hi, can anyone help. Bit correction in previous message, there are no
logs in
This happens mysteriously randomly. If asterisk was killed and
restarted, it often gives this error
myast*CLI core stop now
No such command 'core stop now' (type 'core show help core' for other
possible commands)
Any hint
Thanks
Jim
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