Is it possible to receive a call via IPKall through IAX connectivity without
registration?
If so how to set it up.
I've run-into and old link;
http://forum.voxilla.com/ipkall-support-forum/ipkall-beta-testing-iax-connectivity-without-registration-26728.html
--
Joseph
Hello,
I have a 3 server asterisk configuration where one asterisk (say A) (v
1.4.25) has a digiuim card connected to E1 from which calls are routed
to another asterisk server (B) (1.6.0.9) over SIP trunk from which
calls get routed to third server (C) (1.6.0.9) via IAX trunk.
SIP clients are
hey , all
i have one issue on incoming DAHDI PRI
it works fine many times but sometimes it creates bad audio and also having
echo in line
also recording going to be disturbed by this
i cannot understand this properly
can any one have solutions and how to improve this also how to monitor lines
also sprach Luki lugos...@gmail.com [2009.09.19.0745 +0200]:
sounds like the hiccup my E71 had once. I think the symptoms were
identical. Changing the transport type from Auto to UDP solved the
problem for me. The Auto setting worked, but only sometimes. Maybe
the E65 is similar...
I've tried
Dear Folks,
Anyone knows if Sangoma supports or going to provide support for battery
removal detection on FXO lines?? As Tzafrir said earlier DAHDI supports it,
which is a very nice feature but what about Sangoma?
Regards.
--
M. Shokuie Nia.
___
--
hey paulh,
i think this would not help
because he wants such a dial command which forwards a call to local server
if server_ip is of same server
i have same kind of problem but still dont found proper solution
in,fact i need dialing on IP base in which dialing by using IP address will
send
Get a VOIP headset, Install VOIP client SW, join the global Asterisk
meeting this
Sunday Sept 20, 12N-3P Pacific Daylight Savings Time (UTC-8),
3P-6P Eastern, (7P-10P UTC?)
http://sites.google.com/site/berkeleytip/remote-attendance
Lots of great, exciting new things for Asterisk users,
as we
Martin,
Try to put qualify=yes.
Torintino
Date: Fri, 18 Sep 2009 22:45:05 -0700
From: lugos...@gmail.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] E65 fails registration, soft phone works
Martin,
sounds like the hiccup my E71 had once. I think the symptoms
also sprach Torintino T torinti...@hotmail.com [2009.09.19.1356 +0200]:
Try to put qualify=yes.
I had qualify=2000, but even with the default, the problem prevails.
Thanks for taking the time to reply,
--
martin | http://madduck.net/ | http://two.sentenc.es/
den stil verbessern, das heißt
M Shokuie wrote:
DAHDI supports it, which is a very nice feature but what about Sangoma?
I would suggest you ask Sangoma, they are very responsive.
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary
Safety, deserve neither Liberty nor
Hi,
I'm a noob to Asterisk and this list so I apologize if I'm out of
place with my questions.
I'm planning to utilise Asterisk to build a switchboard (of sorts)
with it and was wondering if this would be feasible (from what I've read
about Asterisk it seems possible). What I would like to
On Saturday 19 September 2009 01:07:54 Rajkumar S wrote:
I have an occasional problem where DTMF is not recognized, ie if
clients type a digit while in menu the system does not register it.
In my C server I saw a log line like this today:
DTMF end '1' has duration 57 but want minimum 80,
FWIW:
From old, old memory, DTMF was 60 ms on, 40 ms off, way back when. With
modern technology, shorter durations could work. Most phones of all types
don't make a standardized tone burst but produce tones only while the button
is pressed. Fast punching will produce short tones.
On the other
Good afternoon gentlemen (and ladies).
A costumer of mine has many servers and each one maps their SIP extensions
to the others via DUNDi. It works like a charm. SIP extensions can only
register at one server, the one they belong to. In case one extension
wants to call other that is
- JR Richardson jmr.richard...@gmail.com escreveu:
Good afternoon gentlemen (and ladies).
A costumer of mine has many servers and each one maps their SIP
extensions
to the others via DUNDi. It works like a charm. SIP extensions can
only
register at one server, the one they belong
On Sat, Sep 19, 2009 at 3:33 AM, M Shokuie sena...@gmail.com wrote:
Dear Folks,
Anyone knows if Sangoma supports or going to provide support for battery
removal detection on FXO lines?? As Tzafrir said earlier DAHDI supports it,
which is a very nice feature but what about Sangoma?
Hello
Hello.
I'd like to know if the two following functionalities are available in
Asterisk.
-1- A stop/wait/halt functionality in the Dialplan. Like:
exten = myexten, n, Halt
where execution of the dialplan would wait indefinitely. I guess a Wait
would be OK, but I'd like this wait to
Hello;
Asterisk voip configuration settings for nonoh did, but somehow could not
manage. I would like your example of working configuration.
Best regards
M.B.
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2009 -
Hi,
I've seen this USB product from Sangoma :
http://www.sangoma.com/products_and_solutions/hardware/analog_telephony/usb_fxo.html
Is it working ok ?
Is it easy to integrate it with Asterisk ?
How would you rate your experience with it ?
Regards
___
19 matches
Mail list logo
