Displaying 20 results from an estimated 2000 matches similar to: "Message Waiting Indication Astersk and kamailio"
2009 Jul 06
1
Asterisk + kamaili MWI(Message waiting Indication)
hello,
Does anyone know about setup Message wait indication between asterisk and
kamailio
my phone are registered on kamailio and voicemail leaves on asterisk
server.
how do i notify to kamailio that 1 message is leaved for you on your
mailbox.
and also i tried all script listed in voip-info.org.
any one know any working method or anybody have some type of setup which may
help me
any help
2010 Jun 29
1
How to Add IP address to SIP Domain
Dear All,
I have Asterisk and Kamailio Configuration.
everything works fine, now the situation is like i have following Dial
pattern in Dialplan.
exten => s,n, Dial(SIP/1002 at glbvoice.com,20,m)
now in my /etc/hosts i have following entry
192.168.1.30 glbvoice.com
then call get forwarded to kamailio and everything is working fine
now question is if i want add one more domain like
2009 Dec 21
1
Asterisk Heartbeat Monitor for Fail safe.
Dear All,
I want to configure Asterisk/Kamailio Like system monitor with Heartbeat
is there any way to monitor Service
If NODE1 is stopped or over loaded then NODE 2 will work and vice verse.
any help appreciated because i m stuck in heartbeat to configure service.
regards
Dhaval
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2010 May 18
1
[ASTERISK-USER] Meetme Recording issue, recording is 2 times Faster then normal recording.
hello All,
i have one issue with Asterisk Meetme Application
i am recording through Meetme channels through option *'r'* and format for
recording a file is '*wav*'
lines used is DAHDI version 2.1.0.4. and asterisk version 1.6.0.5.
i have very strange problem of meetme_recording ,
once conference starts recording file having a *recording is 2x faster *than
normal recording .
2011 Jan 07
3
Call queues on load-balanced asterisks
Hello,
I have been asked to implement the following design:
Load-balanced Kamailio servers handling registrations and routing. Load-balanced asterisk feature servers handling voicemail and other things Kamailio cannot do. Plus several load-balanced gateways, but they are not relevant to my question.
All this is working fine.
I've now been asked to start implementing calling queues, and my
2010 Mar 17
2
Asterisk 1.6.0.5 and app_system FAILED using TRYSYSTEM
Dear All,
i have following CLI error while try to run this command from Dialplan
*TrySystem("DAHDI/45-1", "asterisk -rx "dialplan add extension
1234111,1,Goto(incomingdundi,s,1) into dundilookup"") in new stack
WARNING[32626]: app_system.c:81 system_exec_helper: Unable to execute
'asterisk -rx "dialplan add extension 1234111,1,Goto(incomingdundi,s,1) into
2015 Mar 09
1
PJSIP and Kamailio without registration
Hi,
I want to have Kamailio in front of one or more Asterisk boxes.
I don't think it is necessary for Kamailio and Asterisk to register with
one another. I'd like for PJSIP to recognise Kamailio by its IP address.
I have two boxes, both have public IP addresses, they also have private IP
addresses and can communicate with each other.
I have a Snom phone accessing Kamailio via its
2010 Sep 14
9
Speech To Text on linux with asterisk
Hi,
Is it possible to record say 30 seconds of audio and then have LumenVox
convert to text ?
or any available tool open source for speech to text .
Regards
Dhaval
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2010 May 17
1
new way of asterisk and kamailio (openser) realtime integration
Hello,
I put together a new tutorial about asterisk realtime integration with
kamailio (openser). This time the database used is the one of asterisk,
also call routing logic is controlled by asterisk, here is the link:
http://kb.asipto.com/asterisk:realtime:kamailio-3.0.x-asterisk-1.6.2-astdb
Practically is an easier way to scale starting from existing asterisk
installations.
The other
2009 Aug 31
2
Asterisk Regular expression to validate any phonenumber
Hi
I am using asterisk version 1.6.0.5
I have build up one utility that will fire Originate Action on Manager...
In which, i have define number to call eg. 919912312345 (MobileNumber)
How can i know that this number format is true for Indian Number...
In originate action, user can enter any international number.. How can I
came to know this number format is right for that country...??
IS there
2010 May 17
1
R: new way of asterisk and kamailio(openser) realtime integration
Works for me....
Thanks,
Hristo Benev
-----Original Message-----
From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Alexandru Oniciuc
Sent: Monday, May 17, 2010 6:29 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] R: new way of asterisk and kamailio(openser) realtime integration
2014 Feb 20
2
How to configure asterisk to only accept SIP from kamailio@localhost but exchange RTP on all interfaces?
I have a setup with asterisk-11.7.0 and kamailio-4.1.1. I am following the setup guide at http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb . I want to run asterisk and kamailio on the same server, with SIP realtime configuration
(MySQL database) so that kamailio authenticates and then forwards the registration to asterisk on localhost. The setup calls for asterisk to be
2015 Jan 21
1
PJ SIP realtime with Kamailio / opensips
Hi all,
I saw Matt Jordan's recent Kamailio world talk and was interested in the
idea he proposed of stripping out authentication and registration from
asterisk and letting Kamailio handle it.
All of the tutorials I've seen (e.g. on asipto) show Kamailio forwarding
registrations to asterisk.
In order to do what Matt suggested would I be correct in assuming I would
have to use the
2009 Jul 27
2
Asterisk and Kamailio NAT problem
Hello
Im using Asterisk as a SIP client of Kamailio with RTPproxy. Asterisk is
behind NAT.
X-Lite and SNOM phones behind NAT work fine.
But when I try to connect with an Asterisk behind NAT, the Asterisk
client doesn't receive sound.
I already tried in 2 different NATs, with no firewalls.
This is my Asterisk config:
[kamailio]
type=peer
host=xxx.xxx.xxx.xxx
disallow=all
allow=ulaw
2010 Dec 14
6
Asterisk and Dahdi ON Amazon EC2
Hello Friends,
I am trying to Installl dahdi on amazon EC2 which have Open-SUSE-11.1 X86
version.
and here is snap of uname- a command
*Linux ip-10-160-86-41 2.6.32.19-0.3-ec2 #1 SMP 2010-09-17 20:28:21 +0200
x86_64 x86_64 x86_64 GNU/Linux*
when I try to run DAHDI distribution dahdi-linux-2.1.0.4
I am getting following error
*echo "You do not appear to have the sources for the
2011 Jul 14
9
Extension wise dialplan
Hi all,
I have n no. of extensions in my dialer. from 456 to 556 extensions. I was
created 2 other extensions 667 and 668 I need to allow only STD calls to
go from this extensions.
These all extensions are same context . I need to define the STD dialplan
for only this 2 extensions. how I can ?
Best Regards,
Mahesh Katta
*BUZZ**WORKS* Business Services Private Limited
BANGALORE | CHENNAI |
2015 Jan 29
2
any valid up-to-date info about Kamailio-Asterisk integration ?
Hi all
Have recently watched Matt Jordan's session on Kamailio World 2014
On slides 26-29 of his presentation
(http://www.kamailio.org/events/2014-KamailioWorld/day1/09-Matt.Jordan-Asterisk12-And-PJSIP.pdf)
he speaks about a (completely new, for me at least) approach to build
scalable telephony systems, using N instances of Kamailio and N
instances of Asterisk
Are there any
2013 Jun 28
1
Asterisk behind NAT and Kamailio --> Internal IP in SDP and not "externip"
Hi,
We have some Asterisk servers that we are moving behind a NAT to
preserve public addresses and make room for growth. This is Asterisk 1.4
NAT works very good with the externip/localnet-setting when we are
connected directly to our teleco. But when I try to use NAT and put them
behind our Kamailio something interesting happens: The media-address in
the SDP is the internal ip and not the
2009 Jul 08
3
Asterisk and Skype
Hello All,
can anybody tell me how can i integrate asterisk and skype users
so that skype users can dial my asterisk number or dial internal dialplan
form skype
regars
Dhaval
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2014 Jul 26
1
Rejecting secure audio stream without encryption details - when using ws clients and Kamailio integration
Greetings,
I've noticed a problem that might originate from my Asterisk configuration,
could use a hand in sorting it out. Problem is a 488 response from Asterisk
whenever it gets RTP/SAVPF profile in the SDP.
My current setup has Asterisk Kamailio realtime integration, and Kamailio
uses dispatcher to route calls for Asterisk to handle. Now I have only one
Asterisk, on the same machine as