Hi
I use PRESENCE, PRESENCE_REGINFO and others, with Cisco phones (trying to run
:-)).
The phones are seen as being registered in the PRESENCE, REGINFO work unless
properly.
But I have a question: in the table active_watchers watcher_domain field is the
IP address of the phone and not the doma
Hi,
a new release of Siremis is out - v4.0.0 - web management interface that
is compatible out of the box with Kamailio v4.0.x series.
More details, including link to install tutorial, are available at:
- http://siremis.asipto.com/2013/05/08/siremis-v4-0-0-released/
Regards,
Ramona
Hello:
I followed the step by step guide
(http://kb.asipto.com/asterisk:realtime:kamailio-3.3.x-asterisk-10.7.0-astdb)
that describe the realtime integration
between Kamailio and Asterisk. I have no problem with registration but when I
try a call from 106 to 107 I get the followng error :
Hello:
I followed the step by step guide
(http://kb.asipto.com/asterisk:realtime:kamailio-3.3.x-asterisk-10.7.0-astdb)
that describe the realtime integration
between Kamailio and Asterisk. I have no problem with registration but when I
try a call from 101 to 102 I get the followng error :
when I try a call from 101 to 102 I get the followng error :
1. in the asterisk console: " Unresolvable destination (478/SL)"
2. in the kamailio log:
May 6 17:01:38 WH-PC /usr/sbin/kamailio[4192]: ERROR:
[resolve.c:1540]: ERROR: sip_hostport2su: could not resolve hostname:
"(null)"
Hi all,
currently I have been used kamailio and rtpproxy to listen on two different
network (bridge mode), and the register of sip users are saved on different
locations ("aliases" and "location") according the interface where register
is received. When there is a call to a user registered, I use
hi,
i am from china.i have install a kamailio server with wss support ,and i
set 4443 as the port(set 8080 as the ws port);the result of "netstat -tnl"
is:
"Active Internet connections (only servers)
Proto Recv-Q Send-Q Local Address Foreign Address State
tcp0 0 127
Dear All
I have a Kamalio proxy, and number of phones are connected to it.
When all the phones make call, Kamailio forwards00 request to main proxy
from same source port.
Is there any configuration option in Kamailio.cfg for changing source port
for each call.
Basically one dedicated connection pe
Hello,
The port is configured in the sip.conf file.
Gj
-Original Message-
From: sr-users-boun...@lists.sip-router.org
[mailto:sr-users-boun...@lists.sip-router.org] On Behalf Of zhengyw
Sent: dinsdag 7 mei 2013 11:27
To: sr-users@lists.sip-router.org
Subject: Re: [SR-Users] Σχετ: Problem
On 08.05.2013 04:49, Khoa Pham wrote:
@Olle
"If you have a server on a public IP running behind Kamailio you might
not need RTPproxy relaying for calls to and from that server. Asterisk
will handle NAT by itself and doesn't need help if you turn on NAT
support in Asterisk. In that case, RTPpro
On 5/8/13 10:14 AM, Victor V. Kustov wrote:
Hello, Daniel-Constantin!
Hello,
can you make the behavior configurable via module parameter?
I think yes, with documentation, it took some days.
ok, attach the patch as a file on the mailing list or bug tracker, don't
paste it inline because it
Hello, Daniel-Constantin!
>Hello,
>
>can you make the behavior configurable via module parameter?
I think yes, with documentation, it took some days.
--
WBR, Victor
JID: coy...@bks.tv
JID: coy...@bryansktel.ru
I use FREE operation system: 3.8.4-calculate GNU/Linux
_
Hello,
On 5/1/13 5:08 PM, m...@brightvoip.co.uk wrote:
Hi all,
Posted a similar query a few weeks ago, without much interest - any
advice appreciated.
I have two sites and will send calls between them. I have Kamailio at
each site which will route the calls out/in.
There are multiple distinct
Hello,
can you make the behavior configurable via module parameter? The patch
does not seem to be that intrusive, by just making this option
configurable would be no reason to not accept it, the old style can
still be used. As said previously, I am not using radius myself, just
looked at the
Hello,
look on the wiki for tutorials of installing kamailio (those for debian
should just work on ubuntu). Then read the readme files for msrp and
websocket modules, they have sample config snippets inside.
Cheers,
Daniel
On 5/7/13 3:50 AM, 李启明 wrote:
hi,
i am chinese,i am not good at ka
sip.conf
--
WBR, Victor
JID: coy...@bks.tv
JID: coy...@bryansktel.ru
I use FREE operation system: 3.8.4-calculate GNU/Linux
___
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