Hello,
no time to look at config, but if you run the sip server on a private IP
behind a port forwarding address, you have to use also rtpproxy with
advertising address -- see the second parameter of rtpproxy_manage() or
search on the web for a patch to rtpproxy to add advertising address via
Hello,
rather old, but can still give an idea about config file structure:
- http://kamailio.org/docs/ser-getting-started/SER-GettingStarted.pdf
How to install it, step by step from sources, can be found at:
- http://www.kamailio.org/wiki/install/4.1.x/git
More resources are in the wiki, for
On 02/09/14 19:05, Alex Villacís Lasso wrote:
El 02/09/14 05:17, Daniel-Constantin Mierla escribió:
If you get signling routed ok but no audio, then you have problems
bridging rtp stream.
Most probably you need to use rtpproxy (eventually with advertise
address (there is a patch or use
On Wed, Sep 3, 2014 at 11:36 PM, Daniel-Constantin Mierla
mico...@gmail.com wrote:
Hello,
The $si is the source ip of the sip packet. Bu in a rtimer route there is no
packet received from the network.
So your config is not going to work for what you want to do.
You can try to use mqueue to
On 03/09/14 02:28, James Cloos wrote:
KD == Klaus Darilion klaus.mailingli...@pernau.at writes:
KD Maybe we can find some software with SNI support and BSD license
KD and then copy/paste the code.
nginx is a possibility.
Thanks for the hint. I will have it in my mind if Klaus or some other
Hi Daniel,
Thanks for reply.
I did install patched rtpproxy and did configure it the way you have
described (advertising address - found that after posting the comment). But
it still does not seem to work.
I don't quite know how can i debug, if rtpproxy is actually being used.
Regards,
On 28/08/14 20:33, Tim Chubb wrote:
I'm assuming with the 5080 that this call goes through the Asterisk box before
hitting the registered user on Kamailio... if that's correct, have you also
forced a CALLERID(name) on the call?
A grep of the sip traffic would show if you have something
On 27/08/14 13:08, Satish Patel wrote:
I have post many question on topology hiding any not get any reply back from
people and developers
Many people from the list have their job and lot of them are traveling
quite often, to my knowledge. Therefore some messages can be lost in
archive, so
Hello,
for public sip service at iptel.org use the mailing list at:
- http://lists.iptel.org/mailman/listinfo/services
Cheers,
Daniel
On 24/08/14 13:42, Alexis Walhovd wrote:
Hi
I just created an account with iptel.org http://iptel.org.
But I can not log in to my account through my SIP
Hello,
On 23/08/14 11:47, Olli Heiskanen wrote:
Hello,
A question on Kamailio variables and using dispatcher:
When in failure_route I want to know if the request message was going
to a dispatcher ip or a sip client ip (as in any other than dispatcher
ip), how do I make an if statement for
Hello,
On 22/08/14 09:38, Morten Tryfoss wrote:
Hi,
I wonder about the difference between “used” and “real used” shared
memory in Kamailio.
Each allocated chunk of memory is encapsulated in a fragment that has
additional fields (e.g., to be able to find prev/next fragment). Also,
the
On 22/08/14 04:37, Satish Patel wrote:
I am planing to implement mediaproxy so which NAT module will be good
for media proxy and why?
should i use nat_traversal ro nathelper ? I haven't see any example
people using nat_traversa. most of example i found on google are based
on nathelper so
Hello,
try to use t_newtran() before executing the lua script and from lua use
t_reply() -- I expect it aggregates the responses and should no longer
send the redirect.
If it doesn't work, you can look at using htable to mark that the cancel
was arriving and then don't send reply.
Cheers,
On 04/09/14 09:19, aft wrote:
On Wed, Sep 3, 2014 at 11:36 PM, Daniel-Constantin Mierla
mico...@gmail.com wrote:
Hello,
The $si is the source ip of the sip packet. Bu in a rtimer route there is no
packet received from the network.
So your config is not going to work for what you want to do.
Hello,
On 04/09/14 09:20, Abhishek Saini wrote:
Hi Daniel,
Thanks for reply.
I did install patched rtpproxy and did configure it the way you have
described (advertising address - found that after posting the
comment). But it still does not seem to work.
I don't quite know how can i debug,
Slightly offtopic, but for QA reasons I'm interested in logging RTCP reports
from peers that send them. It looks like rtpproxy-ng might be used to do this
(I'm already using rtpproxy anyway). Is someone already doing this and willing
to share some pointers?
Hi Daniel,
Thanks, i was able to use the command you provided, but did not find the
chunks you have specified(a=nortproxy:yes (iirc)) in the data. Checked by
calling from webrtc client to a desktop client(blink).
When is rtpproxy used though? Kamailio says that it only transmits SIP
signals and
Hello,
maybe you can send to mailing list the output of ngrep so we can look
and check if a rtp relay is used.
If you need to bridge webrtc to classic sip phone, you have to use
rtpengine.
Cheers,
Daniel
On 04/09/14 13:01, Abhishek Saini wrote:
Hi Daniel,
Thanks, i was able to use the
Hello,
On 04/09/14 11:05, Daniel Tryba wrote:
Slightly offtopic, but for QA reasons I'm interested in logging RTCP reports
from peers that send them. It looks like rtpproxy-ng might be used to do this
(I'm already using rtpproxy anyway). Is someone already doing this and willing
to share some
Hi,
Please find attached the output of ngrep for three type of
combinations/connections:
key: Blink is the desktop sip client and ntw means network.
blink2blink_same_ntw_successful
webrtc2blink_same_ntw_failed
webrtc2webrtc_same_ntw_successful
We also need to enable webrtc to classic sip phone
Hi,
I'm trying to fetch attribute list with kamailio ldap_search:
ldap_search(ldap://agents/ou=X,dc=,dc=yy,dc=zz?memberOf?sub?(sAMAccountName=$fU));
ldap_result(memberOf/$avp(s:ldapmemberof1));
xlog(L_INFO, LDAP $fU memberOf = $avp(s:ldapmemberof1));
if (ldap_result_next())
{
On Thursday 04 September 2014 15:22:52 Daniel-Constantin Mierla wrote:
- http://kamailio.org/docs/modules/stable/modules/rtpproxy.html#idp1673992
rtpengine (former rtpproxy-ng) should have it as well, I guess.
Found this before posting, but I could not make any sense of it, $rtpstat with
the
On 04/09/14 18:01, Daniel Tryba wrote:
On Thursday 04 September 2014 15:22:52 Daniel-Constantin Mierla wrote:
- http://kamailio.org/docs/modules/stable/modules/rtpproxy.html#idp1673992
rtpengine (former rtpproxy-ng) should have it as well, I guess.
Found this before posting, but I could not
Take a look here:
http://www.kamailio.org/w/business/
Regards,
Ovidiu Sas
On Thu, Sep 4, 2014 at 5:24 PM, Sharan Harkisoon sha...@sharktek.net wrote:
I am in need of a Kamailio expert that has some availability for consulting
services (remote is fine). Feel free to contact me for details.
On 09/04/14 12:01, Daniel Tryba wrote:
On Thursday 04 September 2014 15:22:52 Daniel-Constantin Mierla wrote:
- http://kamailio.org/docs/modules/stable/modules/rtpproxy.html#idp1673992
rtpengine (former rtpproxy-ng) should have it as well, I guess.
Found this before posting, but I could not
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