[Serusers] ipphone---ser---asterisk---ser---pstn

[Greger V. Teigre]

Hi Iqbal,
You didn't include an ngrep trace of this, so it's hard to say anything.
What I can say is that this is a typical "trombone" issue (when SIP and/or 
RTP goes in a loop). When you do stuff like this, it's really hard to debug 
ser.cfg, as several INVITEs and other messages go back and forth with the 
same call-id...
    I cannot say anything else than that I had been worried too, there is 
definitely something wrong with the signalling.
g-)

Iqbal wrote:
> Hi
>
> I am setting up the following
>
> ipphone registered with ser, the username on ipphone belongs to
> special group called "asterisk"
> when he makes call, I append a prefix, and route his calls to
> asterisk.
> The prefix is appended because I have several "special" groups which
> all hit different extensions in asterisk.
>
> Anyhow this call gets to asterisk, where I then do a few mysql queries
> to find which group/company this user belongs to and then drop them
> into the correct context for their company.
>
> For any of these special users then need to dial 9[number] to route
> the call outside their corporate network, otherwise they can all dial
> internally using 3 digit numbers.
>
> The problem that I have got is that the calls get through okay, and
> both side can talk, but
> a) the call setup takes time
> b) the sip debug just does not seem correct, in fact its got way too
> much going on for my liking.
>
> setup, iphone ------ser------asterisk
>                               |
>                               |
>                            pstn GW
>
> ngrep
> --------
> iphone  invite to SER
> SER ---100 trying ---> ipphone
> SER --- INVITE ---> asterisk
> Asterisk ----> 100 trying ----> SER
> Atsreisk ----INVITE ----> SER
> SER -----100 trying -----> Asterisk
> SER -----INVITE ---PSTN GW
> PSTN GW ----100 trying -----> SER
> PSTN GW ----> 183 session progress ----> SER
> SER ---- 183 session progress ----> Asterisk
> Asterisk ----183--->ser
> ser .-----183 ---> ipphone
> PSTN GW ----200 OK ---> SER
> SER -----200 OK ---->  Ast
> Ast ----ACK ----> SER
> SER ----ACK ---> GW
> AST ----OK ----> SER
> SER ----OK ----IPphone
> ipphone ----ACK ---> SER
> SER ------ACK ---AST
>
> Now around about here is where I think it should stop, cause it all
> seems to make sense...but heres where is starts to go wrong, I then
> get
> Ast ---INVITE ---> ser
> ser ----> 404 User Not found ----> ast
> ast ---ack ---> ser
> ser ---ack ---> gw
> ser ---invite --ast
>
> and various combos of this, but the call is going through. I am
> particularly concerned with the user not found part, since asterisk ip
> is trusted and the call does go through.
>
> Iqbal
>
>
>
>
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