hello all
i have asterisk 1.8.22 and have problem with caller id. this is my
scenario:
PSTN -- FXO --- FXS --- phone(223)
when i call from a 223 to another phone, every thing is ok and caller id
(223) is shown in called phone. but when i call from another phone to 223,
no caller id is shown and
Hi Johan.
But the option maxusers should work too, right?
On Fri, Jul 19, 2013 at 2:52 PM, Johan Wilfer li...@jttech.se wrote:
2013-07-19 15:35, Thiago Coutinho skrev:
Hi all.
I'm trying to limit the number of participants in a conference room
with the realtime option maxusers, but it
On 22/07/2013 5:40 AM, Zoltán Fekete wrote:
Hi!
I have exactly the same problem on asterisk 1.8.22.0 and also on
separate 11.2.1 when sending fax to PSTN.
Tryed with spa-3102, spa-2102, Patton Smartnode 4634, and Zoiper softphone.
SpanDsp also works without any problem on my box.
As I
On Monday 22 July 2013, Josh Hopkins wrote:
Would it be possible to set the ringtone based on the number that was
dialed?
If the phones you are using allow the ringing tone to be changed by sending a
SIP header, yes.
Example of what the goal is:
Dial Denver number
Incoming
Would it be possible to set the ringtone based on the number that was dialed?
Example of what the goal is:
Dial Denver number
Incoming calls ring with ringtone 1
Dial main number
Incoming calls ring with ringtone 2
We are currently using Digium D40, D50, D70 phones.
--
We have Asterisk1.8.11 and can not move to a newer version right now. But when
we run Asterisk as a service, the -r option does not result in giving the CLI
prompt? Did the option to get the CLI change?
--
_
-- Bandwidth and
A quick update.
The nick: theory was proven to be wrong. The incoming calls
consistently fail with or without nick: tag.
I am concentrating on the incoming calls for now.
-Vladimir
On 7/21/2013 3:34 PM, Vladimir Mikhelson wrote:
Hi All:
Has anybody tackled the latest Google Voice issue
Hi all,
I'm not sure how this happened, but one of my customers managed to
turn call forwarding on on his spa112. I thought I had that turned
off in the provisioning file.
I have this in the provisioning file:
Cfwd_All_Serv_1_No/Cfwd_All_Serv_1_
Cfwd_Busy_Serv_1_No/Cfwd_Busy_Serv_1_
And I
Hello
I need to deploy asterisk on production and same thing for DAHDI, which version
is recommended for this?
Regards
Bilal--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for
The a=T38MaxBitRate issue you refer to was one that was actually
discovered at my company and submitted by a colleague. It was fixed in
11.3.0 and 1.8.21.0. However, I think that it wouldn't help based on the
description below being that the parameter was missing altogether. I think
if that
Not sure if this is the right place to mention it, but .
The server downloads.asterisk.org was refusing FTP connections last
night, and
still does not seem to be accepting them this morning.
FTP may not be modern or trendy, but the ability to navigate around
folders textually is
On 23/07/2013 6:18 AM, Kevin Larsen wrote:
The a=T38MaxBitRate issue you refer to was one that was actually
discovered at my company and submitted by a colleague. It was fixed in
11.3.0 and 1.8.21.0. However, I think that it wouldn't help based on the
description below being that the parameter
On 22/07/2013 10:19 PM, Larry Moore wrote:
On 22/07/2013 5:40 AM, Zoltán Fekete wrote:
Hi!
I have exactly the same problem on asterisk 1.8.22.0 and also on
separate 11.2.1 when sending fax to PSTN.
Tryed with spa-3102, spa-2102, Patton Smartnode 4634, and Zoiper
softphone.
SpanDsp also works
If anybody reads this thread here is the solution.
It appeared to be some strange corruption of my Asterisk. As I started
debugging and recompiled everything returned back to normal.
What still puzzles me how some Google Voice accounts continued working
all the time.
-Vladimir
On 7/22/2013
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