result.
Any help would be very appreciated!
Cheers,
Frederic Van Espen
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On Sat, 2012-08-18 at 10:55 +0200, Frederic Van Espen wrote:
Hi,
I just ran into what seems to be an issue on re-invites. I'm not sure if
it's a bug or as designed, so I thought I'd ask the question.
Here's my setup:
- Asterisk 1.8.13.0
- Phone A: Polycom ip331, only allowed to use ulaw
Hi,
I'm in the following usecase:
A customer calls in to an asterisk box and the call is answered by an
employee. During this call, the employee has to set the CDR(accountcode)
for the channel from his phone. Is this possible with asterisk?
What I've tried so far is to add a dynamic feature in
On Thu, 2012-08-30 at 08:23 -0500, Danny Nicholas wrote:
More information please - Asterisk version and are you using realtime?
Currently running asterisk 1.8.13.0 and not using realtime.
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On Sat, 2012-09-01 at 10:05 +0200, Olle E. Johansson wrote:
There is a hidden feature for SNOM phones in the SIP channel. They
have a way to send a client
code during the call (made for lawyers) and that will end up in the
CDR.
That is exactly what we needed Olle. Thanks! The use case is
On Thu, 2012-11-15 at 08:52 +0100, Olivier wrote:
When a Polcyom SoundPoint gets a VLAN ID from a DHCP server, does it
store this VLAN ID into its flash memory so that, on next boot, it
would broadcast its DHCP request using the VLAN he previously got ?
As far as I know it doesn't. It
On Thu, 2012-11-15 at 08:53 +0100, Hans Witvliet wrote:
In stead of 12345678 i would like to use b.c.o.gr...@minoss.nl
But afaicr the dots will cause problems
If your extension does not start with an underscore, it is not
considered as an extension pattern. Correct me if I'm wrong please!
On Tue, 2012-11-20 at 15:03 +0100, gincantalupo wrote:
I'm sure nobody has added something... tried prilocaldialplan and
pridialplan but nothing changed.
Question: if pridialplan or prilocaldialplan would work, should I see
the 0 inside PRI frame with intense debug or it is hidden?
Somebody
Then if you did not restart dahdi and asterisk, then the changes to the
parameters in chan_dahdi.conf and system.conf were never taken into
account. There is no other way than really restarting asterisk and
dahdi.
Frederic
On Wed, 2012-11-21 at 09:08 +0100, gincantalupo wrote:
I cannot restart
On Mon, 2013-03-11 at 14:34 +0100, Patrick Lists wrote:
On 03/11/2013 12:53 PM, termo termosel wrote:
Hi,
I have Ubuntu and Asterisk 11.2.1 in a boot USB. When I put it in
desktop computer, asterisk starts without problem but if I insert
the
same USB in a laptop computer Asterisk
Hi all,
I'm wondering what the recommendations are for using music on hold on
asterisk. As far as I understood from various pages on the web and a
response from the IRC channel, I am to avoid using mp3 files because of
licensing and transcoding issues. correct?
I am currently using asterisk
On 04/23/2013 03:12 PM, Tzafrir Cohen wrote:
If you use that mode, you're probably doing something wrong following an
ancient guide.
Well, these modes are the ones documented in the sample conf files that
came with asterisk 1.8.13.0:
snip
; valid mode options:
; files -- read files from a
Hi,
I think I encountered a bug in the qualify timers for IAX on asterisk
1.8 but I'd like to check if I'm not messing up in my config somewhere
before reporting a bug.
In my IAX peer configuration I have this:
[remote-host]
type=friend
host=172.16.6.45
username=remote-host
secret=test
On 10/02/2013 09:33 AM, s m wrote:
and the last question is how many license key should i buy? i read
that license for g729 is per-channel but i don't understand what channel
exactly means here. this is my scenario :
10endpointspbx181...pbx182...pbx183...10endpoints
pbx181 and pbx183 has
On 10/17/2013 09:47 AM, Alban Elziere wrote:
I'm using Ubuntu server (32bit mainly), standalone or VM (esxi) with good
stability.
Same here. We've been using ubuntu lucid 32bit for years. We have about
1000 implementations of this.
--
Could it be due to the version ubuntu?
Tried to put the asterisk 11 with ubuntu 10.04 - the same error occurs
intermittently.
Try running a memory test on your machine. segfaults can also be caused by
bad RAM. Although you would see other processes than asterisk segfaulting
as well. Do you
On Thu, Apr 3, 2014 at 11:34 AM, A J Stiles
asterisk_l...@earthshod.co.uk wrote:
On Thursday 03 Apr 2014, Павел Чашков wrote:
Could it be due to the version ubuntu?
Tried to put the asterisk 11 with ubuntu 10.04 - the same error occurs
intermittently.
It's possible that this could be a libc
Hi,
I am trying to validate a setup with a sangoma A116 card (16PRI in one
card). I currently have two machines set up, each with a sangoma A116
card. Those are interconnected with crossed PRI cables. One of them is
in NT mode, the other in TE.
The ports are configured in E1 mode, asterisk is
This however gives a never ending flood of output on the pattest on
machine 2. Am I correct in assuming that this is not a good thing?
The errors I was seeing were being caused by the echo cancellation
modules on the A116 card kicking in. All I had to do was disable the
module before starting
Hello list,
We're currently facing some issues concerning T.38 gateway faxing.
This is a device used almost exclusively for receiving faxes. Calls
are incoming to asterisk on a SIP trunk (sangoma netborder) using
G711A. Gateway mode is activated in the asterisk dialplan towards a
Cisco SPA 112
Hi,
On Tue, Dec 2, 2014 at 9:24 AM, Recursive li...@binarus.de wrote:
- Packets 14313, 14314: The provider re-invites asterisk for T.38 (confirmed
by viewing the packet's details), asterisk answers Trying ... to the
provider
- Packets 14315, 14321, 14322: Asterisk re-invites the local
Hi,
On Mon, Dec 15, 2014 at 9:03 AM, Recursive li...@binarus.de wrote:
I would be grateful if you could refer to my message from some minutes ago. I
have provided all the details there.
According to the detailed trace asterisk is indeed retransmitting SIP
OK messages:
snip
Session
Hi,
On Thu, Jan 1, 2015 at 7:09 PM, Recursive li...@binarus.de wrote:
1) Did anybody test T.38 with SPANDSP? If yes, which version of SPANDSP did
you use? Mine is 0.0.6 PRE 20. Should I try to upgrade to PRE 21? Or to one
of the snapshots?
I remember that I was having issues with an older
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