On Sat, 2010-02-13 at 12:54 +0530, cool dude wrote:
hi friend need ur help in dial plan, i want to allow exten 2000 to
2005 can make call outside and exten 2006 to 2010 can not make call
outside. heres my dial plan.
sip.conf
[general]
port = 5060
bindaddr = 0.0.0.0
context = others
12 feb 2010 kl. 16.43 skrev Klaus Darilion:
Am 11.02.2010 21:09, schrieb Olle E. Johansson:
11 feb 2010 kl. 13.30 skrev Klaus Darilion:
Am 11.02.2010 11:21, schrieb Armin Schindler:
Hello,
using Asterisk 1.4.28, I encountered a problem with SIP
RTP port allocation.
I found
On Sat, Feb 13, 2010 at 07:58:39AM +, Brian wrote:
You don't have an extension '9193696136' or anything to handle it. I'm
not familiar with your dialcodes, so assuming you are dialing '9' for an
outside line, followed by a 9 digit number this may work (untested)
exten =
Tzafrir Cohen wrote:
On Sat, Feb 13, 2010 at 07:58:39AM +, Brian wrote:
ZPA/1 rather than ZAP/1-1
That'd be ZAP/1
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary
Safety, deserve neither Liberty nor Safety.
--
Ben Schorr wrote:
Is there some reason why I keep getting this same message from “cool
dude” over and over and over? And under different subject lines?
I assumed that he was sending it to every post, trying to get a
response. Sorta spammy in my opinion.
Doug
--
Ben Franklin quote:
On Sat, 2010-02-13 at 07:03 -0500, Doug Lytle wrote:
Tzafrir Cohen wrote:
On Sat, Feb 13, 2010 at 07:58:39AM +, Brian wrote:
ZPA/1 rather than ZAP/1-1
That'd be ZAP/1
Doug
Thank you Doug - that brightened up my day * 10 - being dyslexic I'd not
spotted that in the first
good question. i never investigated this issue more exact. Any other
T.38 more knowing here if this is possibly anyway?
Kristijan
2010/2/12 Deepesh D deep.d2...@gmail.com:
Hello,
Is it possible to use asterisk in T.38 pass through mode with reinvite?
My fax calls are getting disconnected
On Fri, Feb 12, 2010 at 10:52 AM, Deepesh D deep.d2...@gmail.com wrote:
Hello,
Is it possible to use asterisk in T.38 pass through mode with reinvite?
My fax calls are getting disconnected if canreinvite=yes. It works
only if I make canreinvite=no. Normal calls work in both cases.
Thanks
Hi All,
I read some discussions about the new SIP authentication methods for
1.6.X branches and possible addition of new type of user, type=trunk.
I'm wondering about the disposition about this. Will it be added?
In 1.2 and 1.4 branch, a SIP invite was first checked for a valid
[user] then a
Tim Johnson wrote:
I know this is slightly off topic, but I was wondering if anyone can help
with a problem getting my PAP2's to connect to Asterisk. I use a
provisioning file, and I recently re-wrote the files for each PAP2. I had
a small typo and the PAPs logged it as a corrupt file. I
13 feb 2010 kl. 16.57 skrev JR Richardson:
Hi All,
I read some discussions about the new SIP authentication methods for
1.6.X branches and possible addition of new type of user, type=trunk.
I'm wondering about the disposition about this. Will it be added?
Not to the 1.6 branches, but we
hi i have to create voicemail for exten 2000. below is dial plan which i made.
1 - exten 2000 should receives incoming call from channel
2 - if exten 2000 dosent receives call in 15 secs message should be stored in
voicemail
here r the voicemail,sip, extension.conf
Hi
On Sun, Feb 14, 2010 at 02:29:54AM +0530, cool dude wrote:
hi i have to create voicemail for exten 2000. below is dial plan which i
made.
This message sounds familiar.
In fact, I believe you sent some 20 copies or so of it to the list.
There were actually some useful replies to them.
Friends,
Last week, Hans Petter Selansky alerted us of a potential security issue in all
releases of Asterisk. In fact, it doesn't involve the code, but the most common
way to construct dialplans. If you have something like this in your Asterisk,
you need to update your dialplans:
Excellent and very informative article, Thanks Olle.
I ran thru lots of my dialplans now quickly to see if I have a catch
all exten anywhere. I couldn't find any that are accessible
unauthenticated, I always declare all fixed length extensions using
patterns the exception being international
Much to my surprise I tried to debug an AGI script today with agi
debug on the Asterisk CLI and it did not work. Plus, I could find no
reference on lie of it being removed.
Is there another name for that command? I scanned the CLI help but
found nothing similar. Both my 1.6 boxes do not have the
On Sat, Feb 13, 2010 at 10:50 PM, Alejandro Recarey
alexreca...@gmail.com wrote:
Much to my surprise I tried to debug an AGI script today with agi
debug on the Asterisk CLI and it did not work. Plus, I could find no
reference on lie of it being removed.
Is there another name for that command?
On Sun, Feb 14, 2010 at 04:50:06AM +0100, Alejandro Recarey wrote:
Much to my surprise I tried to debug an AGI script today with agi
debug on the Asterisk CLI and it did not work. Plus, I could find no
reference on lie of it being removed.
Is there another name for that command? I scanned
I'm having problem passing Caller ID to asterisk from AudioCodes MP-114 (FXO)
AudioCodes is passing the caller ID to Asterisk but Asterisk is trying to
interpret it as authentication:
[Dec 31 11:41:07] WARNING[9752]: chan_sip.c:8553 check_auth: username mismatch,
have pstn-5665, digest has
On Sat, Feb 13, 2010 at 09:25:01PM -0500, C F wrote:
Excellent and very informative article, Thanks Olle.
I ran thru lots of my dialplans now quickly to see if I have a catch
all exten anywhere. I couldn't find any that are accessible
unauthenticated, I always declare all fixed length
20 matches
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