Thanks Richard, I will trying to use your solution on practice.
Need roll-back of default FS configs, because it seems that my /dev/hands
not working good)
2014-03-28 20:48 GMT+02:00 Richard Brady :
> Hi Alexandr
>
> On 28 March 2014 15:36, Alexandr Usov wrote:
>
>> I am
2014-03-28 18:16 GMT+02:00 Frank Carmickle :
>
> Freeswitch does not require registration. What are you trying to use
> freeswitch for? Voicemail, B2BUA (transcoding), ivr? See the acl
> configuration or don't use the directory at all.
>
>
I want use Kamailio only for registrations and use rtppro
I am already have some practice to integrate Kamailio with Asterisk, when
all users creates and registers in Kamailio, and calls go to/from Asterisk
with static "host=kamailio_ip" settings for each user on Asterisk side.
I can't (don't know - how to) use in same way integration with FreeSWITCH.
Ca
; ;-)
> Rainer
>
>
> Am 24.03.2014 20:28, schrieb Alexandr Usov:
>
> It is work for qualify - thanks.
>
> Looking for solution for external Asterisk subsribtion of presence states.
> Found the Pinan projec in the web, but it seems only Asterisk 1.4
> supported.
>
014 15:23:31 Alexandr Usov wrote:
> > Peers (from Internet behind NAT) registered on Kamailio (local ip
> > 192.168.182.1), calls from/to routed via Asterisk (192.168.182.24).
> >
> >
> > Can't use qualify info:
> >
> > <--- SIP read from UDP:192.1
dleWatchers 0
It is always Idle, even offline on Kamailio.
So I belived that qualify options can help me, but qualify check only host,
but not user@host
2014-03-24 17:37 GMT+02:00 Daniel Tryba :
> On Monday 24 March 2014 15:23:31 Alexandr Usov wrote:
> > Peers (from Interne
Peers (from Internet behind NAT) registered on Kamailio (local ip
192.168.182.1), calls from/to routed via Asterisk (192.168.182.24).
Can't use qualify info:
<->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '
1efe8f023a80bfb343495d5c4f20ea35@192.168.182.24:5060' Method: O
Kamailio (192.168.182.1) and Asterisk (192.168.182.24) realtime integration
by
http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb.
Kamailio have a public interface also.
Grep from debug on kamailio 4.2.1 on call from asterisk extensions/peers:
Mar 19 14:51:50 netbox /u
Any answer? =(
2014-02-06 Alexandr Usov :
> Integration - works.
> Problem - dialing peer to peer via Kamailio OK but with missing VARs and
> extension number, on dialing/transferring.
> Maybe you know other way to configure Asterisk dialplan for users,
> registered on kamai
Integration - works.
Problem - dialing peer to peer via Kamailio OK but with missing VARs and
extension number, on dialing/transferring.
Maybe you know other way to configure Asterisk dialplan for users,
registered on kamailio and alowing dial as SIP/UserNumber insted SIP/
usernum...@kamailio.host.
Finally it's working:
#!define ASTERISK_LAN1 2.2.2.0/24
# End second LAN for PBXs - tunneling with LAN1 over OpenVPN:
#!define ASTERISK_LAN2 3.3.3.0.0/24
rtpproxy in bridge mode.
And route[NATMANAGE] from Asipto tutorial (
http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-as
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
> 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
}
}
if (is_reply()) {
if(isbflagset(FLB_NATB)) {
fix_nated_contact();
}
}
#!endif
return;
}
2013/8/6 SamyGo
> Please check the rtpproxy function and paste the way it is written in your
> configuration file. Share the output of
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
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<>
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
hen that is not equal to "ei" combination of the flag.
>
> I also suggest that you turn on sip debug on the call receiving asterisk
> and observe the SDP for an incoming call from Kamailio. that will help you
> figure out the situation in SDP.
>
> Best Regards,
> Sammy
&g
even
> how to bridge between ipv4 and ipv4 networks -- you can use it as reference:
>
> - http://kb.asipto.com/kamailio:kamailio-mixed-ipv4-ipv6
>
> Cheers,
> Daniel
>
> On 8/5/13 7:12 PM, Alexandr Usov wrote:
>
>
>
> I have Kamailio on OpenSUSE with static 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,
I have Kamailio on OpenSUSE with static real Public IP (WAN), for ex.
1.1.1.1. I have LAN IP 2.2.2.2. Asterisk as KVM virtual machine with LAN IP
2.2.2.101 and default GW not the SuSe (2.2.2.2), but 2.2.2.1 pfsense LAN
with PUB IP 1.1.1.2) I am configured Registration of UA on Kamailio DB, and
on A
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