18 Jul
2005
18 Jul
'05
8:50 a.m.
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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