e to
0.0.0.0. Why it is happening?
For temporaly solution i had to patch res_config_mysql.c at line 342, added
such lines:
if ((!strcmp(newparam, "ipaddr")) && (!strcmp(buf, "0.0.0.0"))){
ast_log(LOG_DEBUG,"MySQL RealTime: Avoided to updat
Download for Pentium4
Regards,
Mindaugas Kezys
http://www.kolmisoft.com
MOR - Advanced Billing for Asterisk PBX
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of bilal ghayyad
Sent: Wednesday, January 30, 2008 10:55 PM
To: asterisk-users@lists.digium.com
owing way:
User -> Local PBX -> Main GTW -> PSTN
That way you will be save from transfer issue and all your clients will be able
to transfer their calls on Local PBX.
Regards,
Mindaugas Kezys
http://www.kolmisoft.com
MOR - Advanced Billing for Asterisk PBX
-Original Message---
on
person who built this system.
Regards,
Mindaugas Kezys
http://www.kolmisoft.com
MOR - Advanced Billing for Asterisk PBX
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of LWATCDR
Sent: Friday, February 01, 2008 6:14 PM
To: Asterisk Users Mailing List
.,1,Playback(demo-echotest)
exten => _X.,2,Echo()
exten => _X.,3,Hangup
or
exten => _X.,1,Answer
exten => _X.,2,Echo()
exten => _X.,3,Hangup
Regards,
Mindaugas Kezys
http://www.kolmisoft.com
MOR - Advanced Billing for Asterisk PBX
-Original Message-
From: [EMAIL PROTE
Linksys SPA 2102. No issues at all. Period.
Regards,
Mindaugas Kezys
http://www.kolmisoft.com
MOR PRO - Advanced Billing for Asterisk PBX
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Adam Moffett
Sent: Wednesday, February 20, 2008 11:26 PM
To
Cyber-Telecom's CT-V372 is same box as PorTech MV-372 but with more advanced
firmware. It supports more functions, such as SMS sending.
Regards,
Mindaugas Kezys
http://www.kolmisoft.com
MOR PRO - Advanced Billing for Asterisk PBX
-Original Message-
From: [EMAIL PROTECTED] [m
OPENH323DIR=/usr/src/openh323_v1_18_0/
export OPENH323DIR
cd /usr/src/asterisk/channels/h323/
make
make opt
cd /usr/src/asterisk
./configure
make
make install
echo "/usr/local/lib" >> /etc/ld.so.conf
ldconfig
#or similar way
#cp /usr/local/lib/* /usr/lib
Regards,
Mind
We do:
in modules.conf:
noload => pbx_ael.so
noload => pbx_dundi.so
noload => res_config_pgsql.so
noload => res_smdi.so
in extensions.conf delete every context [default], [demo], whatever
in sip.conf, iax.conf delete all peer/users if any
Regards,
Mindaugas Kezys
http://www.kolmi
Linksys SPA942. Tried most of available phones on the market.
These phones sits on companies tables for more then a year.
No problem at all, easy to use, nice(!) to use. I recommend to everybody.
Regards,
Mindaugas Kezys
http://www.kolmisoft.com
MOR PRO - Advanced Billing for Asterisk PBX
Hello,
Just find this file in /var/lib/asterisk/sounds and change it to anything
you like.
Regards,
Mindaugas Kezys
http://www.kolmisoft.com
MOR PRO - Advanced Billing for Asterisk PBX
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Horwich IT
Hello,
Then you can change channel language in front of VoiceMail() app and in
appropriate place put auth-thankyou file which is recorded/made by you.
Regards,
Mindaugas Kezys
http://www.kolmisoft.com
MOR PRO - Advanced Billing for Asterisk PBX
-Original Message-
From: [EMAIL PROTECTED
Hello,
This can help: http://80.86.84.71/kolmiwiki/index.php/Send_Receive_Fax-T38
Regards,
Mindaugas Kezys
http://www.kolmisoft.com
MOR PRO - Advanced Billing for Asterisk PBX
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andreas van
dem Helge
Sent
Hello,
Higher speeds then 9600kbps are not permited by patents.
Regards,
Mindaugas Kezys
http://www.kolmisoft.com
MOR PRO - Advanced Billing Solution for Asterisk PBX
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andreas van
dem Helge
Sent: Friday
Hello,
This can help:
http://www.voip-info.org/wiki/view/Asterisk+Voicemail+ODBC+storage
Regards,
Mindaugas Kezys
http://www.kolmisoft.com
MOR PRO - Advanced Billing for Asterisk PBX
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ein Bielaczyc
Sent
.
Regards,
Mindaugas Kezys
http://www.kolmisoft.com
MOR PRO - Advanced Billing Solution for Asterisk PBX
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Tuesday, March 18, 2008 6:12 PM
To: asterisk-users
Subject: [asterisk-users
lude it into my
application (maybe there's other way?).
I can do like this:
exten => _X,1,Set(PEERIP=${SIPCHANINFO(peerip)})
exten => _X,2,custom_app
and read PEERIP with
pbx_builtin_getvar_helper, but that's not an option for me.
Any help?
Thank you.
Thank you!
You saved my day!
Regards,
Mindaugas Kezys
http://www.kolmisoft.com
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Tilghman Lesher
> Sent: Tuesday, March 25, 2008 5:43 PM
> To: Asterisk Users Mai
Hello,
Rxfax from agx-ags-addons always crashes for us also.
You can download apps we use from:
http://193.138.191.205/packets/fax_apps_asterisk14.tgz
Regards,
Mindaugas Kezys
http://www.kolmisoft.com
MOR PRO - Advanced VoIP Billing
From: [EMAIL PROTECTED]
[mailto:[EMAIL
:2547
2547if (!(hp =
ast_gethostbyname(tmp->value, &ahp)) || (memcmp(&hp->h_addr, &sin->sin_addr,
sizeof(hp->h_addr {
Sorry, I have no time to read manual how to correctly put this into bug
tracker.
Back to 1.4.18.1
Regards,
Mindau
Google is your friend:
http://www.voip-info.org/wiki/view/Asterisk+call+forwarding
Regards,
Mindaugas Kezys
http://www.kolmisoft.com
MOR PRO - Advanced VoIP Billing Solution
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of gilbert
saunders
Sent: Wednesday, April
As far as I noticed - this issue is not 1.4.19 only. Same thing happens on
all Asterisk versions.
Set your own variable before transfer:
Exten => , Set(__MYACC=${CDR(accountcode)})
And use ${MYACC} in other (transfered) calls.
Regards,
Mindaugas Kezys
http://www.kolmisoft.
apps
make
make install
Restart Asterisk.
Voila!
Regards,
Mindaugas Kezys
http://www.kolmisoft.com
MOR PRO - Advanced Billing for Asterisk
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Martin
Sent: Wednesday, April 16, 2008 5:07
Check who is dialing this line by CallerID, if it is not your user - just
drop the call.
Regards,
Mindaugas Kezys
http://www.kolmisoft.com
MOR PRO - Advanced Billing for Asterisk
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of gilbert
saunders
Sent: Thursday, April 17
Hello,
Is here an easy way to get RTCP Stats in channel variables after the call
ends?
Or source should be edited to accomplish this?
I would like to know this before developing this feature.
Regards,
Mindaugas Kezys
http://www.kolmisoft.com
MOR PRO - Advanced Billing for Asterisk PBX
://wiki.kolmisoft.com/index.php/How_fast_MOR_can_perform
It shows how our billing application performs on top of Asterisk (2049
channels) and we can push it even further with some improvements.
We DO NOT RESTART our Asterisk installations daily or weekly. They work for
months.
Regards,
Mindaugas
Make sure /usr/bin/perl can be reached.
Also try in your CLI:
agi debug
Same case happens when I do not have php-cli installed for php AGI scripts.
Mindaugas Kezys
http://www.kolmisoft.com
MOR - Advanced Billing for Asterisk PBX
From: [EMAIL PROTECTED]
[mailto:[EMAIL
exten => _X.,1,Playback(beep)
exten => _X.,2,MixMonitor.
If you are starting the recording using some DTMF code sequence described in
features.conf make sure you use "caller", "callee" or "both" value to play
sound to correct line end.
Mindaugas Kezys
htt
You can always press # at the end of your number to send it to Asterisk.
Mindaugas Kezys
http://www.kolmisoft.com
MOR - Advanced Billing for Asterisk PBX
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jim Houser
Sent: Thursday, November 15, 2007 6:38
arning if 'y' is defined. If
LIMIT_WARNING_FILE is not defined, then the default behaviour is to announce
("You have [XX minutes] YY seconds").
More details about L option for Dial cmd:
http://www.voip-info.org/wiki-Asterisk+cmd+Dial
Mindaugas Kezys
http://www.kolmisoft.c
gests.
Regards/Pagarbiai,
Mindaugas Kezys
Advanced Billing for Asterisk PBX
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Marc LEURENT
Sent: Friday, November 23, 2007 7:04 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Best Pr
For testing purposes you can try one of these:
http://kvin.lv/pub/Linux/Asterisk/
Mindaugas Kezys
http://www.kolmisoft.com
Advance Billing for Asterisk PBX
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Fernando
Berretta
Sent: Friday, November 23
Rename to codec_g729.so
<http://asterisk.hosting.lv/bin/codec_g729-ast14-gcc4-glibc-athlon-sse.so>
Copy to /usr/lib/asterisk/modules
chmod 777 codec_g729.so
restart Asterisk
show translations
Mindaugas Kezys
http://www.kolmisoft.com
Advanced Billing for Asterisk PBX
From:
If you have any questions - there's forum on www.kolmisoft.com/mor to ask
questions and get answers.
Mindaugas Kezys
http://www.kolmisoft.com
Advanced Billing for Asterisk PBX
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Marc LEURENT
Sent: M
your DB on other server, that let's you connect several
Asterisk servers to one DB - unified configuration
Cons:
1. None
Regards/Pagarbiai,
Mindaugas Kezys
Advanced Billing for Asterisk PBX
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alan Lord
Maybe this can help:
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+AMD
Mindaugas Kezys
http://www.kolmisoft.com
MOR - Advanced Billing for Asterisk PBX
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tong
Sent: Sunday, December 02, 2007 7
your great job!
Mindaugas Kezys
http://www.kolmisoft.com
MOR - Advanced Billing for Asterisk PBX
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Johansson Olle E
Sent: Saturday, December 15, 2007 12:57 PM
To: Asterisk Non-Commercial Discussion Users Mailing
Because Realtime Engine is not able to find my Provider which is NOT DYNAMIC!
No other settings changed. Same configuration files. res_config_mysql.so
recompiled to 1.4.16.1.
Please help or explain what's wrong!
Regards,
Mindaugas Kezys
http://www.kolmisoft.com
MOR - Advanced Billing for A
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michiel van Baak
Sent: Thursday, December 20, 2007 12:22 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Realtime logic in Asterisk 1.4.16.1
On 00:12, Thu 20 Dec 07, Mindaugas Kezys
December 2007 16:12:27 Mindaugas Kezys wrote:
> [Dec 20 00:04:12] DEBUG[25686]: res_config_mysql.c:138 realtime_mysql:
> MySQL RealTime: Retrieve SQL: SELECT * FROM devices WHERE name = 'Provider'
> AND host = 'dynamic'
>
> Note: host = 'dynamic'
Correct, t
Do not forget to evaluate Linksys SPA phones. Best I tried and not expensive.
We use them (SPA942) in our company. Everybody's happy.
Regards,
Mindaugas Kezys
http://www.kolmisoft.com
MOR - Advanced Billing for Asterisk PBX
-Original Message-
From: [EMAIL PROTECTED] [mailto:[
model which is
dialing and so on.
This method lets you retrieve call status info from DB without using AMI – thus
not bothering Asterisk at all.
Regards,
Mindaugas Kezys
http://www.kolmisoft.com
MOR – Advanced Billing for Asterisk PBX
From: [EMAIL PROTECTED] [mailto:[EMAIL
s will end more and more people will start to complain about this.
Regards,
Mindaugas Kezys
http://www.kolmisoft.com
MOR - Advanced Billing for Asterisk PBX
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anthony Messina
Sent: Sunday, December 30, 2007 12:09
m could be only
for SIP and less trouble to troubleshoot.
Thanks for info.
Regards,
Mindaugas Kezys
http://www.kolmisoft.com
MOR - Advanced Billing for Asterisk PBX
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anthony Messina
Sent: Sunday, December 30,
Thank you!
Will it come to 1.4.16.3 or 1.4.17?
Regards,
Mindaugas Kezys
http://www.kolmisoft.com
MOR - Advanced Billing for Asterisk PBX
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tilghman Lesher
Sent: Sunday, December 30, 2007 5:07 PM
To
MOR Billing comes in PRO and FREE versions. FREE version is OpenSource. It
supports Realtime and GUI is written in RoR. It has own app_mor.so written
in C and is very fast.
You can check it: http://www.kolmisoft.com/mor
Regards/Pagarbiai,
Mindaugas Kezys
http://www.kolmisoft.com
-Original
That's a good note about MySQL replication. You can use it to remove
point-of-failure which currently is your DB server.
Regards/Pagarbiai,
Mindaugas Kezys
http://www.kolmisoft.com
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of James Collier
ow you use them.
Regards/Pagarbiai,
Mindaugas Kezys
VoIP Billing Solutions
http://www.kolmisoft.com
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Patrick
Sent: Wednesday, August 29, 2007 2:45 PM
To: Asterisk Users Mailing List - Non-Commercial Discussi
You can try MOR FREE billing system for Asterisk.
LiveCD can be downloaded from:
http://www.kolmisoft.com/mor/index.php?option=com_content&task=view&id=73
Regards/Pagarbiai,
VoIP Billing Solutions
Mindaugas Kezys
http://www.kolmisoft.com
-Original Message-
From: [EMAIL P
You can try this: http://www.voip-info.org/wiki-NVFaxDetect
Regards/Pagarbiai,
Mindaugas Kezys
VoIP Billing Solutions
http://www.kolmisoft.com
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Olivier
Sent: Wednesday, August 29, 2007 4:32 PM
To: Asterisk Users Mailing List
You can try MOR: www.kolmisoft.com/mor
It does what you need. It does it even in FREE version.
PRO version costs _many_ times less then other not free solutions mentioned
in this thread.
Regards/Pagarbiai,
Mindaugas Kezys
http://www.kolmisoft.com
From: [EMAIL PROTECTED]
[mailto
Hi,
Try downgrade to Asterisk 1.4.18.1. It works for us perfectly with H323.
Following versions has nasty bugs, not actually related to H323, but who knows,
maybe it will help to downgrade.
Regards,
Mindaugas Kezys
http://www.kolmisoft.com
VoIP Billing and Routing Solutions
interpretation for providers
* Currency engine rework - automatic update from web
Regards,
Mindaugas Kezys
http://www.kolmisoft.com
VoIP Billing and Routing Solutions
___
-- Bandwidth and Colocation Provided by http://www.api
ody experienced same situation? Maybe there is some fix?
Regards,
Mindaugas Kezys
http://www.kolmisoft.com
VoIP Billing and Routing Solutions
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To U
ting:
1. Who/what answers the calls at the other end? I guess real live traffic
should be sent through this Asterisk server?
2. How many calls you had made to to diagnose your problems?
Thank you.
Regards,
Mindaugas Kezys
http://www.kolmisoft.com
VoIP Billing and Routing Solutions
___
bottleneck.
As we achieved our goal - no further testing was performed.
As conclusion - we are very happy with Asterisk in this case.
If somebody is interested - more details are here:
http://wiki.kolmisoft.com/index.php/Asterisk_Background_performance_test
Regards,
Mindaugas Kezys
Hello,
We made small stress-test for H323.
Test shows that H323 protocol is heavyweight compared with SIP.
More details: http://wiki.kolmisoft.com/index.php/H323_pass-through_test
Regards,
Mindaugas Kezys
http://www.kolmisoft.com
VoIP Billing and Routing Solutions
Not sure about OOH323, but H323 can:
http://www.voip-info.org/wiki/view/Asterisk+RealTime+H323
Regards,
Mindaugas Kezys
http://www.kolmisoft.com
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Bruce Ferrell
> Sent: Sa
05/packets/asterisk1.4.21_noresponse.jpg
Back to 1.4.18.1 (1.4.19.x is even more broken:
http://lists.digium.com/pipermail/asterisk-users/2008-April/209342.html).
Regards,
Mindaugas Kezys
http://www.kolmisoft.com
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michael J.
Liber
Hello,
Anybody can advice how to pass variable between 2 Asterisk servers over
IAX2?
With SIP I can use SipAddHeader.
How do to the same with IAX2?
Thank you.
Regards,
Mindaugas Kezys
http://www.kolmisoft.com
___
-- Bandwidth and Colocation
Thank you.
Regards,
Mindaugas Kezys
http://www.kolmisoft.com
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Tilghman Lesher
> Sent: Friday, June 27, 2008 6:33 PM
> To: Asterisk Users Mailing List - Non-Commer
Hello,
We are proudly to present new version of our billing and routing system MOR
v0.6
More info: http://www.voip-info.org/wiki/view/MOR
Regards,
Mindaugas Kezys
http://www.kolmisoft.com
___
-- Bandwidth and Colocation Provided by http://www.api
Hi,
Try to delete whole column 'md5secret' from DB peers table.
Leave only 'secret'. And try then.
Regards,
Mindaugas Kezys
http://www.kolmisoft.com
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of
Hello,
Based on our own and our clients' experience we compiled short manual: How
and to whom sell VoIP
Hope it can be useful to some of you also.
You can download it from our site: http://www.kolmisoft.com
Regards,
Mindaugas Kezys
http://www.kolmisof
12921
Also how do you go about changing details for device in DB and not using
"sip realtime prune PEER" + 'sip reload'?
Without that your changes to devices are not active.
Good luck!
Regards,
Mindaugas Kezys
http://www.kolmisoft.com
> -Original Message-
> Fr
call
first after call was answered?
Thank you.
Regards,
Mindaugas Kezys
http://www.kolmisoft.com
VoIP Billing and Routing Solutions
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To
How to determine which channel hung up first?
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 Geoff Lane
Sent: 2009 m
call first.
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 Costa Tsaousis
Sent: 2009 m. vasario 21 d. 17:30
To: asterisk-users@lists.digiu
!
/etc/init.d/asterisk start
# clean
rm -rf /tmp/f1
rm -rf /tmp/f2
Regards,
Mindaugas Kezys
http://www.kolmisoft.com
VoIP Billing and Routing Solutions
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Any guidelines how to solve locked channels problems?
E.g. to find out which part of the code has problems and causes locks.
Upgrade to newer versions are not an option.
Regards,
Mindaugas Kezys
http://www.kolmisoft.com
VoIP Billing and Routing Solutions
-Original Message-
From
Locked channel does not react to 'soft hangup' command.
That's why it is called - LOCKED.
Regards,
Mindaugas Kezys
http://www.kolmisoft.com
VoIP Billing and Routing Solutions
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:aste
Check this:
http://www.voip-info.org/wiki/index.php?page=Call+Quality+Metrics
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
Sorry for replying for the second time, but this issue is interesting for me
also.
I found such link: http://www.nessoft.com/kb/50
And this: http://www.jdsu.com/product-literature/voipstats_an_acc_tm_ae.pdf
Regards,
Mindaugas Kezys
http://www.kolmisoft.com
VoIP Billing and Routing Solutions
Could you share with us your Openoffice callc function?
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 Marc Leurent
Sent
Formula here: http://www.nessoft.com/kb/50 has jitter in it.
Regards,
Mindaugas Kezys
http://www.kolmisoft.com
VoIP Billing and Routing Solutions
-Original Message-
From: Marc Leurent [mailto:lf...@leurent.eu]
Sent: 2009 m. balandžio 2 d. 13:56
To: asterisk-users@lists.digium.com
Cc
Try this:
Exten =>_X.,n,Dial(SIP/ser_sei0/1130,,L(10208400:61000:1))
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 BERG
session 3ad367ee48778d2c523a60e62ae86...@85.113.41.129
And media still goes through Asterisk.
Why is that?
Why strict routing is enforced?
Regards,
Mindaugas Kezys
http://www.kolmisoft.com
VoIP Billing and Routing Solutions
s from our experience and is
totally unusable.
We ended making #exec based script which takes data from DB and forms static
configuration on each reload.
Regards,
Mindaugas Kezys
http://www.kolmisoft.com
VoIP Billing and Routing Solutions
-Original Message-
From: asterisk-
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.digiu
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
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
027277...@prov7|60|S(7197)'
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
) Duplicate
entry '' for key 1
This happens when you have UNIQUE for some field and insert same record twice.
In order to help please paste your [cdr] table structure.
I'm sure it's not a bug but misconfiguration which can be solved easily.
Regards/Pagarbiai,
Mindaugas Kezys
http
sterisk+local+channels
And do your originate like this:
ACTION: Originate
Channel: Local/[EMAIL PROTECTED]/n
Exten: 4321
Priority: 1
Context: blabla
And in
[blabla]
Exten => 4321,1,Dial(Local/something/n)
I guess you got the idea.
Regards/Pagarbiai,
Mindaugas Kezys
http://www.kolmisoft.co
Hi,
Connect you Asterix box in the middle of call-flow and you will be able to
record all calls.
PSTN <-> Asterisk <-> Legacy PBX <-> Phones
Regards/Pagarbiai,
Mindaugas Kezys
http://www.kolmisoft.com
VoIP Services&Solutions
MOR - FREE Open Source billing for Asteri
* Callback
* Click2Call
* Invoice generation
* Custom Rates
* Localization
More info in: http://www.kolmisoft.com/mor
Regards/Pagarbiai,
Mindaugas Kezys
http://www.kolmisoft.com
___
--Bandwidth and Colocation provided by Easynews.c
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 Solu
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...@list
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
1 - 100 of 125 matches
Mail list logo
