18 Jan
2004
18 Jan
'04
10:08 p.m.
Hi !
After specific time I redirect (revert_uri and append_branch)
call to another sip address. Everythig is ok for UA like ATA, Kphone
and C7960). When the call is started from Grandstream
after the pick up second site (Asterisk IVR- after redirection),
connection is terminated afer a few seconds.
This situations takes place (only for GS ) also when I redirect calls from
one sip domain to another depends on prefix call (for client
doesn't support URL sip addresses like GS, ATA)
In logs ACK message directed to ser I see differences between UA.
Originated destination sip address is 3000, when no answer, call is
redirected to 4000
For ATA I have:
192.168.0.83:5060 -> 192.168.0.1:5060
ACK sip:3000@192.168.0.1 SIP/2.0..
Route: <sip:4000@192.168.0.81;branch=0>,<sip:4000@192.168.0.1:6060>..
Via: SIP/2.0/UDP 192.168.0.83:5060..
From: radan <sip:3100@sip.router.pl;user=phone>;tag=3207317092..
To: <sip:3000@sip.router.pl;user=phone>;tag=as2d60db53..
Call-ID: 3934861712(a)192.168.0.83..
CSeq: 1 ACK..
User-Agent: Cisco ATA 186 v3.0.0 atasip (031210A)..
Content-Length:
0....
For GS I have:
192.168.0.84:5060 -> 192.168.0.1:5060
ACK sip:4000@192.168.0.1:6060 SIP/2.0..
Via: SIP/2.0/UDP 192.168.0.84..
Route:<sip:3000@192.168.0.1;ftag=f6e3b058-8afd-fac2-e60b-e493a7d83844;lr>..
Route: <sip:4000@192.168.0.81;branch=0>..
From: "radan - grandstream"
<sip:3102@sip.router.pl;user=phone>;tag=f6e3b058-8afd-fac2-e60b-e493a7d83844..
To: <sip:3000@sip.router.pl;user=phone>;tag=as18e54868..
Contact:<sip:3102@192.168.0.84;user=phone>..
Call-ID: 683b3311-0ebd-55c5-4191-5469b058efb1(a)192.168.0.84..
CSeq: 65090 ACK..
User-Agent: Grandstream SIP UA 1.0.3.81..
Max-Forwards: 70..
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, INFO, SUBSCRIBE..
Content-Length: 0....
Two different calls are confirmed.
for GS I have then following info a few times (5 or 6)
192.168.0.1:5060 -> 192.168.0.84:5060
SIP/2.0 200 OK..
Via: SIP/2.0/UDP 192.168.0.84..
Record-Route: <sip:4000@192.168.0.81;branch=0>..
Record-Route: <sip:3000@192.168.0.1;ftag=f6e3b058-8afd-fac2-e60b-e493a7d83844;lr>..
From: "radan - grandstream"
<sip:3102@sip.router.pl;user=phone>;tag=f6e3b058-8afd-fac2-e60b-e493a7d83844..
To: <sip:3000@sip.router.pl;user=phone>;tag=as18e54868..
Call-ID: 683b3311-0ebd-55c5-4191-5469b058efb1(a)192.168.0.84..
CSeq: 65090 INVITE..
User-Agent: Asterisk PBX..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER..Contact:
<sip:4000@192.168.0.1:6060>..
Content-Type: application/sdp..
Probably GS is not able to send ACK
After them the ser sends BYE to the GS
192.168.0.1:5060 -> 192.168.0.84:5060
BYE sip:3102@192.168.0.84;user=phone SIP/2.0..
Record-Route: <sip:3000@192.168.0.1;ftag=as18e54868;lr>..
Max-Forwards: 9 ..
Via: SIP/2.0/UDP 192.168.0.1;branch=z9hG4bK2743.08055687.0..
Via: SIP/2.0/UDP 192.168.0.81;branch=z9hG4bKcc8e.cc7088e2.0..
Via: SIP/2.0/UDP 192.168.0.1:6060;branch=z9hG4bK0ff02add..
From: <sip:3000@sip.router.pl;user=phone>;tag=as18e54868..
To: "radan - grandstream"
<sip:3102@sip.router.pl;user=phone>;tag=f6e3b058-8afd-fac2-e60b-e493a7d83844..
Contact: <sip:4000@192.168.0.1:6060>..
Call-ID: 683b3311-0ebd-55c5-4191-5469b058efb1(a)192.168.0.84..
CSeq: 102 BYE..User-Agent: Asterisk PBX
Content-Length: 0....
a GS talks that this connection doesn't exist
192.168.0.84:5060 -> 192.168.0.1:5060
SIP/2.0 481 ..
Via: SIP/2.0/UDP 192.168.0.1;branch=z9hG4bK2743.08055687.0..
Via: SIP/2.0/UDP 192.168.0.81;branch=z9hG4bKcc8e.cc7088e2.0..
Via: SIP/2.0/UDP 192.168.0.1:6060;branch=z9hG4bK0ff02add..
Record-Route: <sip:3000@192.168.0.1;ftag=as18e54868;lr>..
From: <sip:3000@sip.router.pl;user=phone>;tag=as18e54868..
To: "radan - grandstream"
<sip:3102@sip.router.pl;user=phone>;tag=f6e3b058-8afd-fac2-e60b-e493a7d83844..
Call-ID: 683b3311-0ebd-55c5-4191-5469b058efb1(a)192.168.0.84..
CSeq: 102 BYE..
User-Agent: Grandstream SIP UA 1.0.3.81..
Content-Length: 0....
It is a some bug in soft for GS, or do I have to
add something special in configuration file for GS ?
Thanks
Andrzej