We use 1.4.18.1 and 1.4.26.1 and it does not work - settings are not changed
after prune, asterisk must be reloaded, sip reload or iax2 reload makes
changes.
But after that all devices loose registration.
Regards,
Mindaugas Kezys
http://www.kolmisoft.com
VoIP Billing and Routing Solutions
(see Section 6.5.1). If a participant
generates multiple streams in one RTP session, for example from
separate video cameras, each MUST be identified as a different
SSRC.
Regards,
Mindaugas Kezys
http://www.kolmisoft.com
VoIP Billing and Routing Solutions
-Original Message
Asterisk sometimes goes to sleep. (And never wakes-up).
Restart it and all will be fine again.
We have a watchdog which sends SIP OPTIONS packet to Asterisk and if it does
not respond – restarts it.
Regards,
Mindaugas Kezys
http://www.kolmisoft.com
VoIP Billing and Routing Solutions
Check this link: http://wiki.kolmisoft.com/index.php/RTPAUDIOQOS_Demystified
Regards,
Mindaugas Kezys
http://www.kolmisoft.com
VoIP Billing and Routing Solutions
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of DHAVAL
INDRODIYA
Thank you for answer. It was very informative, I put it in our wiki if you
don't mind.
Regards,
Mindaugas Kezys
http://www.kolmisoft.com
VoIP Billing and Routing Solutions
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com
We had many problems with IAX2, changing to SIP solved them all.
Let me paste link to wise-words which clearly illustrates our experience:
http://wiki.kolmisoft.com/index.php/Why_we_do_not_suggest_to_use_IAX2
Regards,
Mindaugas Kezys
http://www.kolmisoft.com
VoIP Billing and Routing Solutions
it will be useful for starters and makes life easier for many
people.
Link to get it:
http://www.kolmisoft.com/billing-and-routing/mor-voip-billing-demo/
More info about software: http://www.voip-info.org/wiki/view/MOR
Regards,
Mindaugas Kezys
http://www.kolmisoft.com
VoIP Billing and Routing
Try:
killall -9 safe_asterisk
killall -9 asterisk
/etc/init.d/asterisk start
Regards,
Mindaugas Kezys
http://www.kolmisoft.com
VoIP Billing and Routing Solutions
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf
simultaneous calls which is enough for
majority of startups.
You can check our manual to see what functionality is supported:
http://wiki.kolmisoft.com/index.php/MOR_Manual
Regards,
Mindaugas Kezys
http://www.kolmisoft.com
VoIP Billing and Routing Solutions
-Original Message-
From
)'
Call log is here: http://pastebin.ca/1667975
Why Asterisk decided to terminate the call?
Regards,
Mindaugas Kezys
http://www.kolmisoft.com
VoIP Billing and Routing Solutions
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com
existing registrations and all previously
registered devices will be unreachable till they register again.
Regards,
Mindaugas Kezys
http://www.kolmisoft.com
VoIP Billing and Routing Solutions
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun
and try to
retrieve data I need from internal structures using custom c module and
Asterisk API?
I'm trying to retrieve ${CHANNEL(rtpqos,audio,all)} for Leg B.
Regards,
Mindaugas Kezys
http://www.kolmisoft.com
VoIP Billing and Routing Solutions
Just remember, that after reload you will lose all registrations.
Regards,
Mindaugas Kezys
http://www.kolmisoft.com
VoIP Billing and Routing Solutions
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Carlos
Please try our billing which has easier managing interface and works ok with
H323: http://www.voip-info.org/wiki/view/MOR
FREE version is available over this link:
http://www.kolmisoft.com/billing-and-routing/mor-voip-billing-demo/
Regards,
Mindaugas Kezys
http://www.kolmisoft.com
VoIP Billing
The problems we have with Asterisk Realtime:
1. After reload all registrations are void.
2. Without reload prune does not take effect.
Test it in your scenario also.
Regards,
Mindaugas Kezys
Kolmisoft UAB
VoIP Billing Solutions
e-mail: i...@kolmisoft.com
URL: http
Sip reload
Regards,
Mindaugas Kezys
Kolmisoft UAB
VoIP Billing Solutions
e-mail: mailto:i...@kolmisoft.com i...@kolmisoft.com
URL: http://www.kolmisoft.com http://www.kolmisoft.com
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com
From my experience prune does not take effect without reload.
And after reload ALL your phones are unreachable for 2 minutes!
Imagine you have several thousands devices unreachable for 2 minutes.
How much calls will fail during that time?
Regards,
Mindaugas Kezys
Kolmisoft UAB
VoIP Billing
There is no problem to get necessary data for MOS calculation for Call Leg
A using:
${CHANNEL(rtpqos,audio,all)}
How to get similar data for Call Leg B?
It would be very nice to have such info even if it will not lead to correct
MOS calculation.
Regards,
Mindaugas Kezys
Kolmisoft UAB
VoIP
Execute such commands with cronjob every night:
/etc/init.d/asterisk stop
sleep 3
killall -9 safe_asterisk
killall -9 asterisk
/etc/init.d/asterisk start
Regards,
Mindaugas Kezys
Kolmisoft UAB
VoIP Billing Solutions
e-mail: i...@kolmisoft.com
URL: http://www.kolmisoft.com
From: asterisk
in
Asterisk world.
We have 470 servers deployed around the world with Asterisk and this piece
of code extended my and mine coworkers lifes by many years.
If you really want to solve your problem - start here:
http://www.voip-info.org/wiki/view/Asterisk+debugging
Regards,
Mindaugas Kezys
Kolmisoft UAB
Try this: http://www.b2bua.org/wiki/AsteriskCodecNegotiationPatch
Regards,
Mindaugas Kezys
Kolmisoft UAB
VoIP Billing Solutions
e-mail: i...@kolmisoft.com
URL: http://www.kolmisoft.com
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun
1082
Regards,
Mindaugas Kezys
Kolmisoft UAB
VoIP Billing Solutions
e-mail: i...@kolmisoft.com
URL: http://www.kolmisoft.com
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Philipp von
Klitzing
Sent: Tuesday
From our experience it is not enough. We had to rewrite CDR generation to
suite our billing needs. That was on 1.4.xx, we are not using 1.6+
Regards,
Mindaugas Kezys
Kolmisoft UAB
VoIP Billing Solutions
e-mail: i...@kolmisoft.com
URL: http://www.kolmisoft.com
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-Original
We use it to determine who is the caller.
Regards,
Mindaugas Kezys
Kolmisoft UAB
VoIP Billing Solutions
e-mail: i...@kolmisoft.com
URL: http://www.kolmisoft.com
Find us on Facebook
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun
Just use uniqueid, which is exactly what you want. No modification is
necessary.
Regards,
Mindaugas Kezys
Kolmisoft UAB
VoIP Billing Solutions
e-mail: i...@kolmisoft.com
URL: http://www.kolmisoft.com
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