in kamailio 4.0, what could cause these kind of error messages:
Aug 2 09:31:04 voip /usr/sbin/sip-proxy[1779]: ERROR: usrloc [ucontact.c:842]:
db_update_ucontact_ruid(): updating record in database failed - empty ruid
Aug 2 09:31:04 voip /usr/sbin/sip-proxy[1779]: ERROR: usrloc [urecord.c:363]:
Hi daniel,
Thanks for your prompt reply.
How could i add the repository by commands to sources.list?
(i dont know much about linux)
Could you help me on teamviewer to configurate the server so i could learn
to do it myself?
wheezy is for debian64? (as i read other tutorials about squeeze version)
On Tuesday 06 August 2013 16:20:16 william anderson wrote:
> Someone could help me in setup my kamailio vps server?
> So as to learn how to install it (i am newbie).
>
> *Installation is on a VPS server running debian64, so i need to do it all
> by commands.
Add the kamailio.org repository to /et
My bad: "cwii" - sounds ok, but WAN Kamailio in RTP debug "from/to".
rtpproxy_manage("cwie"); - good for Echo() test,
when UA behind NAT, registered on Kamailip and calling Asterisk Echo() test
exten - voice perfect.
RTP - from/to only Kamailio LAN IP (2.2.2.2) - what is my goal.
But need to know
I am gound flags for rtpproxy_manage(which help external (behind NAT) UA
registerd on Kamailio call to Echo() test extension on Asterisk - it is
"cwie". But for Peer-to-Peerm, registered on Kamailio and working though
Asterisk dialplan, it must be rtpproxy_manage("cwii").
Thanks to author of this
hi,
Someone could help me in setup my kamailio vps server?
So as to learn how to install it (i am newbie).
*Installation is on a VPS server running debian64, so i need to do it all
by commands.
Any step-by-step guide, or possible to view how you install it on
teamviewer is perfect so as to learn
Ahh, right! Got it.
Thanks!
-Original Message-
From: sr-users-boun...@lists.sip-router.org
[mailto:sr-users-boun...@lists.sip-router.org] On Behalf Of Alex Balashov
Sent: Tuesday, August 6, 2013 3:40 PM
To: sr-users@lists.sip-router.org
Subject: Re: [SR-Users] Workings of the kamailio d
On 08/06/2013 09:00 AM, Grant Bagdasarian wrote:
Hello,
Consider the following Kamailio script:
route {
….
route(DISPATCH);
route(RELAY);
}
route[DISPATCH] {
ds_select_domain("1", "8");
return;
}
Dispatcher Table
SetID
Dest
> I wonder who this belongs to : c=IN IP4 192.168.144.101
This is Asterisk LAN IP (just nit changed by me before postin here to
2.2.2.101 for better reading WAN/LAN table).
> Also your Kamailio just sends the c=IN IP4 1.1.1.1 for the very first
incoming call that tells me that RTP proxy function
Hello,
Consider the following Kamailio script:
route {
route(DISPATCH);
route(RELAY);
}
route[DISPATCH] {
ds_select_domain("1", "8");
return;
}
Dispatcher Table
SetID
Destination
1
192.168.1.10
1
192.168.1.11
Algorithm 8 uses the fi
Ok thanks,
All fine except the rtpproxy_manage function is just out in the open and
since you're in bridging mode you need to realize that your kamailio may
receive calls from the WAN Interface or from the Asterisks on LAN
interface. So if you're to bridge the RTPs at the proxy then how can
Kamail
ps -ef | grep rtpproxy
kamailio 15853 1 0 12:41 ?00:00:01 /usr/sbin/rtpproxy -u
kamailio -l 1.1.1.1 2.2.2.2 -s unix:/var/run/rtpproxy.sock
netstat -pln|grep rtpproxy
unix 2 [ ACC ] STREAM LISTENING 238561268
15853/rtpproxy /var/run/rtpproxy.sock
I'm trying rtpp
Please check the rtpproxy function and paste the way it is written in your
configuration file. Share the output of "ps -ef | grep rtpproxy" and
"netstat -pln|grep rtpproxy"
--
Sammy
On Tue, Aug 6, 2013 at 3:55 AM, Alexandr Usov wrote:
>
>
> Note:
> Asrterisk LAN IP real 192.168.144.101 but mus
Note:
Asrterisk LAN IP real 192.168.144.101 but must be 2.2.2.101 in this
described network (I am missed to change before copy-pasting here).
___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-router.org
http:/
Hi again,
Still Missing 200OK for this call. It'll be helpful to send a complete
trace for the call coming in to the Asterisk at first place and then
Dialing out to the B-leg whose trace which you've just shared.
On Tue, Aug 6, 2013 at 3:22 AM, Alexandr Usov wrote:
>
>
> <>
> Di
<>
Dial (...) in new stack
== Using SIP RTP CoS mark 5
Audio is at 19614
Adding codec 13 (ulaw) to SDP
Adding codec 12 (gsm) to SDP
Adding codec 14 (alaw) to SDP
Adding codec 100017 (testlaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting
You should not change the kamailio.cfg for nat=yes param, that works the
way it is. Yes you're right changing the NAT param in asterisk won't change
anything.
Please enable sip debug on asterisk and paste the complete INVITE/200OK
packets for the established call with no audio.
--
Sammy
On
It seems I am undesrtand whereis problem can be found.
Original tutorial of Kamailio+Asterisk realtime integration (by Asipto)
containse settings for cheking if the "nat=yes" presents, but in Asterisk
11 I am using nat=force_rport,comedia.
# RTPProxy control
route[NATMANAGE] {
#!ifdef WITH_NAT
Hi Camille,
currently mediaproxy-ng only does timeout notifications only towards a
SEMS-SBC, not towards Kamailio. :-(
Adding support for notificating Kamailio on RTP-Timeout could probably
be added as well, but is not yet available.
Kind regards,
Carsten
2013/8/6 Camille Oudot :
> Hi Carsten,
>
Hi Carsten,
Thanks for the response,
> i wrote the patch quite long ago (2years+?). Back then, i posted the
> patch both on the rtpproxy-devel list and i even asked, if there was a
> tracker for patches; but i never ever received ANY reply... :-(
Ok, I've successfully applied it on an older vers
Dear Alexandr,
You can connect Kamailio to RTPproxy via socket as well, use modparam like
this:
modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:12221")
Then if your rtprpoxy is started in bridged mode you should use the "i" and
"e" flags while you call the rtpproxy-manage() function in the
Thank you for response!
A little difficult for me to find the same logic in my case with tutorial
of ipv4/ipv6 bridgin...
When I started
/usr/sbin/rtpproxy -u kamailio -l 1.1.1.1/2.2.2.2 -s udp:127.0.0.1 12221
There is no sound.
Is this a major to connect via unix sock?:
modparam("rtpproxy", "r
Hi!
I an starting learning syntax from this url:
http://www.kamailio.org/wiki/cookbooks/4.0.x/core
and some actual setting s from:
http://kamailio.org/docs/modules/4.0.x/
I am new with Kamailio)
2013/8/6 刘日新
> Hi, all.
>
> ** **
>
> I has download the updated kamailio with version 4.0,
Hi, all.
I has download the updated kamailio with version 4.0, and I has correctly
configure and install it with some default functions. It did well. Thanks
kamailio.
But when I want customize some features like TLS etc, I got stuck, and I
google them and I just found some notes on the older v
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