when one rtpproxy server is down.
Does anyone tell this solution by the parameter or somthing ?
I am looking forward from anyone.
Thank you so much.
Noriyuki Hayashi
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for pgpool2postgres that is
working all and fine.
I do not know the reason.
By the way I will report to the ALL.
Thank you for kamailio and Daniel, ALL!
Noriyuki Hayashi in Japan
On Mon, 13 Jun 2016 11:03:19 +0200
Daniel-Constantin Mierla wrote:
> Hello,
>
> to understand p
appriciated.
Thank you.
#!define DBURL "postgres://asterisk:password@192.168.192.143:/kamailio"
#!ifdef WITH_ASTERISK
#!define DBASTURL
"postgres://asterisk:password@192.168.192.143:/asterisk"
#!endif
#!endif
Noriyuki Hayashi Japan.
___
same context.
In this case, Kamilio is using default context transfer is aborted.
If anyone has any idea with much appriciated.
Thank you.
Noriyuki Hayashi Japan
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SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users
ll-ID: 799baa2d555fc65c192593e043164160@192.168.192.92:5080.
CSeq: 102 OPTIONS.
Via: SIP/2.0/UDP 192.168.192.92:5080;branch=z9hG4bK5775add0;rport=5080.
Supported: 100rel,replaces,timer.
Allow: INVITE,ACK,OPTIONS,CANCEL,BYE,REFER,NOTIFY,PRACK,UPDATE.
Content-Type: application/SDP.
Content-Length:266.
.
llo,
>
> can you get the ngrep output on kamailio server? From asterisk log I see that
> an INVITE with To-tag has no Route header, which should be there if run
> though kamailio.
>
> Cheers,
> Daniel
>
> On 06/01/14 08:50, Noriyuki Hayashi wrote:
> > Hello,
>
Hello,
I am beginner using kamailio with much appreciated.
Only one sip-phone is hang up after 60 seconds problem.
This sip phone has no nat function at all.(SANYO SIP-2100)
Grand Stream is works fine with kamailio.
I would like give me your great advice with much appreciated.
Environment.
CentOS
