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
___
-- Bandwidt
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 wrote:
> Hi, can anyone help. Bit correction in previous message, there are no
> logs in /var/log/asterisk/messages too.
>
> Thanks
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 wrote:
> Hi,
>
> We are having a strange issue. If we run asterisk from /etc/inittab
> and then connect using asterisk -r, we don't se
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 c
Julian Lyndon-Smith ha scritto:
> If I have the following in the dialplan
>
> exten => foo,n,Dial(SIP/1234&Zap/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 i
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 ch
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
___
-- Bandwidth and Colocation Provided
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 Sat
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
> this post is attached to the prevoius post, this is what i have on CLI
> when i call from Linksys pap2t to aste
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 messag
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
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 usie
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 ext
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?
>
>Phil
On Fri, Feb 13, 2009 at 2:37 PM, John Todd 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/
> >
>
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
Proceeding
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
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 us/8801773
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