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: core
[resolve.c:1540]: ERROR: sip_hostport2su: could not resolve hostname:
(null)
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 :
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
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
FWIW
http://siremis.asipto.com/pub/downloads/siremis/siremis-4.0.0.tgz
Returns 404
..
Thx.
On 5/9/13 2:51 AM, Elena-Ramona Modroiu wrote:
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,
Fixed - a typo in file name.
Thanks,
On 5/9/13 9:07 AM, David | StyleFlare wrote:
FWIW
http://siremis.asipto.com/pub/downloads/siremis/siremis-4.0.0.tgz
Returns 404
..
Thx.
On 5/9/13 2:51 AM, Elena-Ramona Modroiu wrote:
Hi,
a new release of Siremis is out - v4.0.0 - web management
Hello.
As I understand t_load_contacts() / t_next_contacts() can help only in
case when branches have different q values.
The only way I see is to get all contacts via reg_fetch_contacts(), save
the list to an AVP and then iterate through the list in the way
described in tm module docs.
Dear Kamailio experts
I have a typical use case where I want Kamailio to behave as a B2BUA.
What I mean here is (assume Kamailio is using TCP for SIP call
establishment)
1. For each call it should create a separate TCP connection with next proxy
in path.
2. When call ends, it should close that