I had a look at the files and they are really a nightmare to parse. Some
are Word and some are Excel. Good luck :)
l.
2012/3/29 Markus unive...@truemetal.org
http://www.itu.int/oth/T0202.aspx?parent=T0202
But don't do it. Because I'm doing it right now. So let's not waste energy
and do the
On 30 Mar 2012, at 10:14, Syco wrote:
Finally the problem is: I cannot manage more than 80 concurrent calls.
What happens on the 81st call?..
S
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New to
Check the sip.conf.sample file. I think it is the call-limit parameter that
is getting you. The sample file should tell you what the default is.
Another possibility is that your rtp range is set too low; the normal
range is 1-2, which allows for 2500 calls(or 5000 if you set other
Asterisk says to process the call correctly:
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Executing [17000@sipp:1] Answer(SIP/sipp-005a, ) in
new stack
-- Executing [17000@sipp:2] Set(SIP/sipp-005a, rn=100)
in new stack
--
Warren Selby wrote 29.03.2012 22:46:
To do this, you
change your features.conf setting like so:
parse =
*9,peer/both,Macro,Parse
The same result when I changed to Macro. I
believe that it's true that callerid on outgoing call is crap shoot.
Here is output:
-- Executing
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New to Asterisk? Join us for a live introductory webinar every Thurs:
Be it meetme or confbridge, asterisk 1.2.x or Asterisk 10.x. Is it possible
to query a channel and get its conference number in return?
On Thu, Mar 22, 2012 at 11:09 AM, Satish Barot satish4aster...@gmail.comwrote:
Jayesh, Personally I haven't worked on Congbridge :).
Confbridge has evolved a
Core show channels verbose provides this information. Just grep for the
channel you need to hit.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of [Digital^Dude]
R
Sent: Friday, March 30, 2012 7:45 AM
To: Asterisk Users Mailing List -
On 30 Mar 2012, at 10:04, Sean McMaster wrote:
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
It's not tricky.. Really.. It's on the bottom of every email.--
Ok, this was a stupid thing (my fault), with -r 1000 I get easily 1000
concurrent calls that terminate in 20 seconds.
This calls just answer, play a file the first 2 seconds and then wait.
Then sipp close because of two many errors, this is the log:
sipp: The following events occured:
We use MeetMe with res_timing_dahdi as the timing source, and once a while we
get the following error which then causes Asterisk to crash/restart (with safe
Asterisk).
ERROR[6518] res_timing_dahdi.c: Failed to configure DAHDI timing fd for 0
sample timer ticks
According to the following
http://www.linuxsystems.it/2012/03/new-t-38-gateway-patch-against-asterisk-1-8-11-0/
I made a new patch from irroot's branch and I ported it to 1.8.11.
Unfortunately latest one is still against 1.8.8 and porting from
subversion is quite time consuming, hopefully my work will be useful to
On Fri, Mar 30, 2012 at 8:43 AM, Mikhail Lischuk mlisc...@itx.com.uawrote:
**
Warren Selby wrote 29.03.2012 22:46:
To do this, you change your features.conf setting like so:
parse = *9,peer/both,Macro,Parse
The same result when I changed to Macro. I believe that it's true that
On Fri, 30 Mar 2012, [Digital^Dude] ® wrote:
Be it meetme or confbridge, asterisk 1.2.x or Asterisk 10.x. Is it
possible to query a channel and get its conference number in return?
(I'm a 1.2 Luddite.)
Not directly.
You can execute an AGI which can connect to Asterisk via AMI and execute
Try adding
resetinterval = never
here. The default is to reset any idle channels every 3600 seconds.
Unfortunately, if a channel is being reset just as an incoming call
arrives, there is a chance that the channel will get stuck in a
resetting state and block any further use of that
Warren Selby wrote 30.03.2012 18:37:
A couple things - what
version of asterisk are you using? Are you actually using zaptel or do
you have DAHDI as your interface to your TDM cards?
Asterisk
SVN-branch-1.4-r359615
I have to work with wicked Tormenta cards with
weird chip which requires
Hi all,
Does anyone know of any providor that offers free calling to the US?
Feel free to contact me off list.
Many thanks,
Christian--
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Can anybody please tell me any ENUM test DID from e164.arpa tree, which I
can use to test some features?
Thanks,
Ricardo.
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What does this patch fix? Why is it not in Jarr?
Thanks
Bryant
From: Niccolò Belli darkba...@linuxsystems.it
Sent: Friday, March 30, 2012 11:19 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] T.38 gateway patch against Asterisk 1.8.11.0
On Fri, Mar 30, 2012 at 1:16 PM, Bryant Zimmerman brya...@zktech.comwrote:
What does this patch fix? Why is it not in Jarr?
Thanks
Bryant
It looks like the patch is a backport of the t.38 gateway functionality in
Asterisk 1.10.
Ryan
--
Hello nice group,
having a Problem with CDRs.
If i change the context with Goto() Asterisk write the new exten in dst cdr
field.
How can i keep the old entry? Any ideas makes me very happy.
Thanks for helping me.
Daniel
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More information please - 1.8X or 10.X?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Daniel Knoll
Sent: Friday, March 30, 2012 1:17 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
Hi Danny,
Thank you for answer.
I'm using Asterisk 1.8.7.0
Daniel
Am 30.03.2012 um 20:18 schrieb Danny Nicholas da...@debsinc.com:
More information please - 1.8X or 10.X?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
So you have a situation like so:
[default]
Exten = _X.,1,Answer
Exten = _X.,n,Goto(foo,s,1)
[foo[
Exten = s,1,playback(vm-goodbye)
Exten = s,n,hangup()
And you get two CDR records, 1 with default and 1 with foo?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
In your case i have 1 record and only the last one s is in the field dst, but
i don't use s, i use numbers instead.
Thats my situation :(
Am 30.03.2012 um 21:21 schrieb Danny Nicholas:
So you have a situation like so:
[default]
Exten = _X.,1,Answer
Exten = _X.,n,Goto(foo,s,1)
[foo[
Doesnt google voice offer that?
On Fri, Mar 30, 2012 at 1:00 PM, Christian christia...@runbox.com wrote:
Hi all,
Does anyone know of any providor that offers free calling to the US?
Feel free to contact me off list.
Many thanks,
Christian
--
On Fri, Mar 30, 2012 at 2:21 PM, Danny Nicholas da...@debsinc.com wrote:
So you have a situation like so:
[default]
Exten = _X.,1,Answer
Exten = _X.,n,Goto(foo,s,1)
[foo[
Exten = s,1,playback(vm-goodbye)
Exten = s,n,hangup()
And you get two CDR records, 1 with default and 1 with foo?
Il 30/03/2012 19:29, Ryan Wagoner ha scritto:
It looks like the patch is a backport of the t.38 gateway functionality
in Asterisk 1.10.
Yes it's a backport from asterisk 10, asterisk 1.8 does not have the t38
gateway functionality and there is no chance to get t38 gw in 1.8 at
this point.
Il 30/03/2012 19:16, Bryant Zimmerman ha scritto:
Why is it not in Jarr?
Ok I linked it in jira, anyway I don't know how many peoples still
follow the old bug report considering it has been closed.
Niccolò
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Hi,
I want a provider that uses SIP, I live outside of the US.
Many thanks,
Christian
On 2012-03-30 at 15:49 C F wrote:
Doesnt google voice offer that?
On Fri, Mar 30, 2012 at 1:00 PM, Christian christia...@runbox.com wrote:
Hi all,
Does anyone know of any providor that offers free calling
Looks nice, was also my first idea, but field dst is read only. I can't
overwrite this and get an error like this
ERROR[2474]: cdr.c:345 ast_cdr_setvar: Attempt to set the 'dst' read-only
variable!.
Am 30.03.2012 um 22:00 schrieb Warren Selby:
On Fri, Mar 30, 2012 at 2:21 PM, Danny Nicholas
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