Displaying 20 results from an estimated 4000 matches similar to: "Extra REGISTER messages sent by Asterisk when subscribe for MWI is defined in zoiper"
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
2014 Aug 11
1
Letting rtp profiles be handled by rtpengine instead of Asterisk
Hello,
I'm trying to get calls working between websocket clients and sip clients.
For clients I have sip.js based clients on chrome, Zoipers and a
Grandstream phone. Challenge here is I'd like to have Kamailio and
rtpengine to handle the bridging between different rtp profiles but
Asterisk changes them in the sdp bodies along the way. I'm using Asterisk
11.11.0.
Is there a way to
2015 Feb 06
4
Question regarding custom announcements used by several Asterisk servers
Hello,
Got a question regarding custom announcements in Asterisk.
My goal is to allow my users record their own queue announcements and
choose which announcements they want to use in each queue. I have several
Asterisk servers and a Kamailio server which dispatches call traffic
between the Asterisks. Question is, is it possible to have something like a
NSF disk shared between several asterisk
2014 Apr 24
1
Realtime integration: Unregistered clients showing as registered?
Hello all,
I've been testing a Kamailio Asterisk Realtime integration, and found a
strange situation.
My problem is that when using the integration, everything seems ok but
Asterisk does not see the clients as registered. Kamailio and the clients
report registered clients. Also calls fail.
In Asterisk cli sip show peers shows nothing but for example realtime load
sipusers name 660 shows the
2014 Aug 06
1
From and To headers contain same account in INVITEs
Hello,
I noticed a strange thing while testing my Asterisk-Kamailio Realtime
setup. In an INVITE the From and To headers contain the same number when
calling through a Realtime integration setup. This happens when the INVITE
leaves Asterisk.
Can you guys tell me what might be causing this? I have 660 at testers.com as
a websocket client and 700 at testers.com (caller) using a Zoiper client (db
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
2014 Dec 05
2
Inbound call from sip peer to internal webrtc peer fails while internal sip-webrtc calls work
Hello,
I'd appreciate your comments on the following problem I'm having, please
forgive me if this is something obvious, I've been scratching my head on
this for a while:
I have Asterisk+Kamailio setup where I'm currently testing inbound calls
from outside. I have both webrtc and sip clients, where webrtc peers are
defined according to sip.js instructions (
2015 Jan 03
2
Asterisk removes a charachter from sip peer name
Hello all,
Just wondering on a behavior I noticed while testing with realtime sip
peers with names like 111.222 at mydomain.com. Using Kamailio as outbound
proxy, it sends Asterisk a sip message where To header value is <
sip:111.222 at mydomain.com> and From header has value "username" <
sip:111.333 at mydomain.com;transport=UDP>;tag=fc609171. When Asterisk sends
out the
2009 Jun 24
1
Message Waiting Indication Astersk and kamailio
hi all,
I have Asterisk 1.6.0.5 Installed and kamailio 1.0.5 version installed
when i leave voicemail On Asterisk i need MWI Indication on kamailio
extension
there are some methods i tried but still cant get success
All other feature are working fine also try voip-info.org methods
can anybody suggest me for different method and have some different setting
on SIP .
any help appreciated
2010 Apr 15
0
Regarding remote registration of SIP user on zoiper
Hello list,
I am new to this list and have been using Asterisk as part of my research project for about 2 weeks now.
I would like to get your thoughts on a scenario that I am attempting at the moment. I haven't had luck until now.
In this scenario, I am trying to register a SIP user configured on the zoiper client installed on a laptop, which is on the same Local Area Network, with the
2007 Oct 30
2
zoiper iax registation: "facility rejected"
I'm trying to setup zoiper ( formerly idefisk ) to use my asterisk
server at work from home.
I've setup zoiper for iax, set the ip address to work's fixed ip
address, user: home, password: password
but the zoiper log shows:
11:02:35 Rejected registration for 'home@<my-office-ip-address>' with
cause 'facility rejected'
11:03:35 Rejected registration for
2007 Feb 14
0
Zoiper softphone version 1.03 now available
Hello guys!
We released a new ZoIPer BIZ BETA (version 1.03).
You can experience better look and more advanced features. Finally MS
Vista fans can also make use of it.
Zoiper BIZ BETA is available free of charge from www.zoiper.com.
There you can find out more about the improvements and features.
We are also offering customization packages for ZoIPeR Free Windows.
Zoiper is a multiprotocol:
2015 Mar 04
2
WebRTC phone
For those that were interested I have attached the kamailio.cfg which we
have working with Kamailio 4.2.1 and Asterisk 1.8.23/32. Specifically, the
following yum packages:
kamailio.x86_64 4.2.1-4.1
@home_kamailio_v4.2.x-rpms
kamailio-auth-ephemeral.x86_64 4.2.1-4.1
@home_kamailio_v4.2.x-rpms
kamailio-bdb.x86_64 4.2.1-4.1
@home_kamailio_v4.2.x-rpms
2007 Jul 19
1
Idefisk softphone - official 2.0 release - Zoiper
Hello guys,
The so expected 2.0 release of Idefisk 2.0 softphone is a fact.
Idefisk and Zoiper became one - Zoiper 2.06.
Here are some of the features: SIP and IAX, TCP, TLS support,
Multi-language support, Automatic provisioning (XML), URL handling,
Outlook Integration, Native conferencing, API, Changeable number of
lines....
You could read the complete Press Release here:
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 Jul 25
0
Audio Delay of 1-2 seconds, one way with Zoiper soft phones
Hi All
I will do a test call from a soft phone to my mobile. I can speak into my
headset and the audio is heard instantly. But if I speak into my mobile
there is a 1-2 second delay in the Audio. I am using SIP.
I am only finding it in the Zoiper Softphones that we are using. All other
phones don't seem to have it present. Sadly the customer is Quite attached
to the Zoiper.
I have set QOS =
2008 Feb 05
3
[Softphones] ZoIPer vs. XLite?
Hello
I need to hook up someone's remote PC onto our Asterisk server over
the Net. There are firewalls on each side, so I figured it's time to
give IAX a try, and see if it's less of a pain to use than SIP. And
since IAX hardphones are pretty are, I guess I'll go softphone.
Apparently, the two most well-known IAX and SIP clients for Windows
are ZoIPer and X-Lite, respectively.
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
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