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 To :  Vihar Patel <vihar.patel@y...>
 From :  Martin Provencher <mprovencher86@g...>
 Subject :  Re: [yate] Need Assistance to configure Yate for SIP-H323 Proxy Setup
 Date :  Mon, 4 Oct 2010 11:20:08 -0400
Hi Vihar,
      Some answers on your questions :
1. If your using SIP INFO, you should not have any problem at all. If you
are using RFC2833, you need to enable or disable it in ysipchan.conf. Verify
with your SIP side which option you need.
2. I never see this case before. You should verify in a ethernet capture if
your message on the SIP side are well formatted.
3. You will need to understand how regexroutes.conf works to be able to do
that. See http://yate.null.ro/pmwiki/index.php?n=Main.RegularExpressions to
start.

I hope it will help you.

Martin

On Mon, Oct 4, 2010 at 8:59 AM, Vihar Patel  wrote:

> Hi,
>
>
>
> I am trying to set up Yate as a proxy between H323 – SIP network. Where my
> calls are originated from H.323 and SIP routes them. This works with certain
> practical issues which holds me to complete my testing, those issues are
> listed as below:
>
>
>
> 1.       DTMFs are not getting transmitted between both ends – means
> caller (h323) is not receiving any DTMF from its called party which is via
> SIP network or vice versa.
>
> 2.       If called party (via SIP network) terminates the call, caller
> (h323) doesn’t get any termination signal so it keeps the call live.
>
> 3.       I need to understand routing based on different country codes.
>
> E.g.
>
> a.       If caller (h323) wants to dial to USA, call will have 001 and SIP
> provider’s IP will be x.x.x.x.
>
> b.      If caller (h323) wants to dial to UK, call will have 044 and SIP
> Provider’s IP will be y.y.y.y.
>
>
>
> Please help me out to set up this configuration and complete my testing so
> I can move ahead.
>
>
>
> Thanks in Advance!!
>
>
>
> Regards,
>
> Vihar Patel
>



Hi Vihar,
      Some answers on your questions :
1. If your using SIP INFO, you should not have any problem at all. If you are using RFC2833, you need to enable or disable it in ysipchan.conf. Verify with your SIP side which option you need.
2. I never see this case before. You should verify in a ethernet capture if your message on the SIP side are well formatted.
3. You will need to understand how regexroutes.conf works to be able to do that. See http://yate.null.ro/pmwiki/index.php?n=Main.RegularExpressions to start.

I hope it will help you.

Martin

On Mon, Oct 4, 2010 at 8:59 AM, Vihar Patel <vihar.patel@y...> wrote:

Hi,

 

I am trying to set up Yate as a proxy between H323 – SIP network. Where my calls are originated from H.323 and SIP routes them. This works with certain practical issues which holds me to complete my testing, those issues are listed as below:

 

1.       DTMFs are not getting transmitted between both ends – means caller (h323) is not receiving any DTMF from its called party which is via SIP network or vice versa.

2.       If called party (via SIP network) terminates the call, caller (h323) doesn’t get any termination signal so it keeps the call live.

3.       I need to understand routing based on different country codes.

E.g.

a.       If caller (h323) wants to dial to USA, call will have 001 and SIP provider’s IP will be x.x.x.x.

b.      If caller (h323) wants to dial to UK, call will have 044 and SIP Provider’s IP will be y.y.y.y.

 

Please help me out to set up this configuration and complete my testing so I can move ahead.

 

Thanks in Advance!!

 

Regards,

Vihar Patel