I have problem with Asterisk.
[sendCommand]=EXEC DIAL IAX2/somehost/somenumber|10
[readReply]=200 result=-1
[sendCommand]=GET VARIABLE ANSWEREDTIME
1086422 [Thread-7] ERROR - establishConnection: exec encountered exception
net.sf.asterisk.fastagi.AGIHangupException: Channel was hung up.
This means
channel info after hangup).
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Stefan
Reuter
Sent: Saturday, April 01, 2006 2:04 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] AGI hangup problem
Branko Samardzic wrote
Hi everyone,
I am running two Asterisk servers on two machines that have dynamic DNS due
to ISP changing IP address daily. Both servers are registered on DynDns.org
and IP update scripts work fine on both machines. However, if one machine
changes IP address, other one (that has trunk pointing to
It is not problem with outdated DNS cache. It takes approximatelly 2 minutes
for DynDns entry to propagate and my router picks it up. However Asterisk
does reverse name lookup (instead of trying to resolve server.dyndns.org to
IP address and pick up new one, it does reverse lookup on old IP
That's fine. But there is, obviously, situations where such situation is not
welcome. Is it possible to force Asterisk to refresh cache every in a while.
Regards,
Branko
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and you will have DynDns working nicely.
Cheers,
Branko
-Original Message-
From: Branko Samardzic [mailto:[EMAIL PROTECTED]
Sent: Sunday, December 11, 2005 3:25 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Asterisk Dynamic DNS
No problem.
Cheers,
Branko
-Original Message
Hi,
I am experimenting with trunkfreq parameter.
When it is 20ms I can see both parties in IAX session sending IAX frames
every 20ms.
When I modify this parameter to 40ms then I can see that only server that
initiated
IAX connection works properly (i.e. sends IAX frames every 40ms while other
Hi,
I am wondering if it is possible to tweak IAX2 protocol to packetize audio
data
more efficiently. I would like to try setups where multiple audio frames
(gsm)
are combined into single UDP packet. I know that it will incur delay in
audio
streams but I don't care. Primary concern is to lower
Hi Andrew,
thanks for your prompt response. However, I am not sure whether IAX trunking
can be of any benefit on 33.6kbps link. It shows significant bandwith
reduction
on 2 and more simultaneous calls. My question relates to single call over
such
link.
Current measurement say that gsm call
Hi,
I am trying to make following setup
PSTN -- Asterisk_1 --- IAX2 stream over Internet - Asterisk_2 --
PSTN
^
|
AGI control App
Basically this is calling card solution that accepts calls from PSTN of one
country
checks user credentials (PIN and
I am trying to tweak my Asterisk servers to talk to each other using Speex
codec.
I downloaded and installed speex and speex devel libraries, recompiled
asterisk (including make clean), did set up speex codec as only one allowed
on both sides. Sounds enough.
However, conversations are not Speex
I am playing around with different codecs between 2 * servers. However I
don't seem to have any impact on bandwith.
I always get something like this:
Name: IAX2/ds02-1
Type: IAX2
UniqueID: 1131421484.2
Caller ID: s
Caller ID Name: (N/A)
DNID Digits: (N/A)
;hanguponpolarityswitch=yes
Regards,
Branko Samardzic
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] TDM400P hangup detection on Bell Canada
PSTN
On Saturday 05 November 2005 23:22, Branko Samardzic wrote:
I am pretty new to * but it seems that hang-up detection is hot topic.
Does
anyone know how to setup my * for proper detection of hang-up condition.
PSTN lines I use for testing are just plain
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