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
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
Serusers mailing list serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
I actually left out the ngrep cause it was really messy, so I went through and gave a summary, but it definantly resolved itself when I changed the canreinvite, and kept asterisk in the media path, will have to debug some other time, I will leave it for now since it works :-)
Iqbal
Greger V. Teigre wrote:
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
Serusers mailing list serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
.
-
Greger V. Teigre -
Iqbal