[Serusers] SER-routing-problem

[Vlad Costea]

On 7/25/07, Weiter Leiter <bp4mls at googlemail.com> wrote:
>
> Vlad,



SER is only handling signaling, SIP. It is unable to handle "voice packages"
- RTP in this case - by itself. That's why two other - equivalent - helping
applications exist: rtpproxy and mediaproxy.

So you are saying that I should also install these other two applications (
are they applications ? excuse me , but i'm not familiar with this and I dit
not quite undestood what you ment to say)

So, in your scenario with SER is the session established OK? In other words,
is the call established and lasts on both phones for more than 30s (or
more), or less than that: the callee picks up, but this is never perceived
at caller?

No the session is not established with SER  (it gives me busy tone) ;
without SER (from the AS directly to the HT), the phone rings more than 30
sec , but when it is picked-up ... nothing.
And this is  my problem ,that with SER the phone doesn't ring and I do not
know how to make this happen from the ser config file.
If it helps , here is the dial-peer from the AS router to send the packages
to SER:
 dial-peer voice xxxxx voip
 description tech support
 huntstop
 destination-pattern .%00111111T
 redirect ip2ip
 voice-class codec 11
 session protocol sipv2
 session target ipv4:193.226.xxx.xxx       ----> the ip of the computer with
SER installed
 dtmf-relay sip-notify
-I'm sure that this dial-peer is corect because , as i said before , it
worked with a hardware SipServer.

As last resort, do a network sniffing on the SER box and send it over.

This will take some time , because I will have to install Ethereal on linux.



On 7/25/07, Vlad Costea <vlad.costea at interpoint.ro> wrote:
> >
> > No , they are on different networks, and NAT has nothing to do with it
> > because there is no router to do so between them, only the default gateways
> > ( both HT and AS have public ip adresses) . As i said in the first mail , my
> > only problem is the config of SER to receive the voice packages from one IP
> > adress and send them to another IP adress and only that ( something like :
> > listen on : 193.226.xxx.xxx,5060 , send to: 193.230.xxx.xxx ).
>
>
> SER is only handling signaling, SIP. It is unable to handle "voice
> packages" - RTP in this case - by itself. That's why two other - equivalent
> - helping applications exist: rtpproxy and mediaproxy.
>
> Actually I'm not even sure that SER cand replace the hardware version of a
> > Sip Server . The path I described
> > (phone->AS5350->SipServer->HandyTone->phone) ,  works on  an  request/reply
> > system  and  the codec  negotiation  is  made after the reply from the HT;
> > if the reply message is not transmited on the same path there is no codec
> > negotiation , there-for no voice.
>
>
> So, in your scenario with SER is the session established OK? In other
> words, is the call established and lasts on both phones for more than 30s
> (or more), or less than that: the callee picks up, but this is never
> perceived at caller?
>
> Anyway , thanks for trying to help me. Unfortunately, day after day , I'm
> > begining to think that there is no solution for my problem.
>
>
> From my experience, unless running SER, no other (hardware) box can do as
> much as SER&friends can. You just need to clearly say what hurts.
> As last resort, do a network sniffing on the SER box and send it over.
>
> WL.
>
> On 7/25/07, Weiter Leiter <bp4mls at googlemail.com> wrote:
> > >
> > > Are the AS and HT both part of the same network?
> > > Otherwise, most comon diagnosis for your symtom is NAT and you might
> > > find some inspiration as instructed below, or any other NAT support resource
> > > you find on www.iptel.org
> > > ftp://siprouter.onsip.org/pub/gettingstarted/configs/nat-rtpproxy.cfg
> > >
> > > ftp://siprouter.onsip.org/pub/gettingstarted/configs/nat-mediaproxy.cfg
> > >
> > >
> > >
> > > On 7/24/07, Vlad Costea < vlad.costea at interpoint.ro> wrote:
> > >
> > > > Hello.
> > > > I have just installed SER on RedHat 9 Linux with the purpose to
> > > > route voice packages between a PSTN Gateway AS5350 and a Grandstream
> > > > HandyTone 286.Let me tell you how it should work: I have assigned a
> > > > phone number, that when it is picked-up, the IVR from the AS5350 router
> > > > respods and after the press of a key it sends the packages to the HandyTone
> > > > and from there to a normal phone; So far so good because it partialy works,
> > > > that means that the phone rings butt there is no voice. I have managed to
> > > > solve this problem using a hardware Sip Server between the two devices ( in
> > > > this way both HT286 and AS5350 act as clients), but this is not possible any
> > > > more because it was not my server; so i have tried with a software solution
> > > > and I've installed SER. As I said before , I haven't managed to configure
> > > > it to work as I wish and this is why I'am asking for your help; it would
> > > > realy help me if you can provide a ser.cfg example that would do
> > > > just the routing part (no ack no authentification , no mysql,...., just
> > > > routing).
> > > > Please excuse my bad english.
> > > > Thank you very much !
> > > >
> > > > _______________________________________________
> > > > Serusers mailing list
> > > > Serusers at lists.iptel.org
> > > > http://lists.iptel.org/mailman/listinfo/serusers
> > > >
> > > >
> > >
> > >
> > > --
> > > "C is a language that combines all the elegance and power of assembly
> > > language with all the readability and maintainability of assembly language."
> > >
> >
> >
> >
>
>
> --
> "C is a language that combines all the elegance and power of assembly
> language with all the readability and maintainability of assembly language."
>
>
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