client1 <> provider <> kamailio <> asterisk <> client2
when client1 calls client2, everything seems to work. sounds works. after a few seconds the call gets terminated.
after tcpdump syslog and config checks i figured out that asterisk sends 200 OK through kamailio to the provider, after this the provider sends a ACK to kamailio, kamailio however sends this ACK to itself and not to asterisk.
I added the config, syslog and wireshark output (converted)
Hi! Check "did=" in your dialog messages? if not present, use match_dialog_mode = 0.
2014-06-10 18:16 GMT+06:00 Gijs Kwakkel kwakkel1000@gmail.com:
client1 <> provider <> kamailio <> asterisk <> client2
when client1 calls client2, everything seems to work. sounds works. after a few seconds the call gets terminated.
after tcpdump syslog and config checks i figured out that asterisk sends 200 OK through kamailio to the provider, after this the provider sends a ACK to kamailio, kamailio however sends this ACK to itself and not to asterisk.
I added the config, syslog and wireshark output (converted)
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
there is a did field at the first packets (invite) till its 200 OK after that the DID field is gone. i however do not use the dialog module
loadmodule "dialog.so" modparam("dialog", "dlg_match_mode", 0)
gave some errors. should i still try to get the dialog module working?
2014-06-10 17:46 GMT+02:00 pavel@eremina.net eremina.net@gmail.com:
Hi! Check "did=" in your dialog messages? if not present, use match_dialog_mode = 0.
2014-06-10 18:16 GMT+06:00 Gijs Kwakkel kwakkel1000@gmail.com:
client1 <> provider <> kamailio <> asterisk <> client2
when client1 calls client2, everything seems to work. sounds works. after a few seconds the call gets terminated.
after tcpdump syslog and config checks i figured out that asterisk sends 200 OK through kamailio to the provider, after this the provider sends a ACK to kamailio, kamailio however sends this ACK to itself and not to asterisk.
I added the config, syslog and wireshark output (converted)
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Can you show here this errors? 12.06.2014 13:37 пользователь "Gijs Kwakkel" kwakkel1000@gmail.com написал:
there is a did field at the first packets (invite) till its 200 OK after that the DID field is gone. i however do not use the dialog module
loadmodule "dialog.so" modparam("dialog", "dlg_match_mode", 0)
gave some errors. should i still try to get the dialog module working?
2014-06-10 17:46 GMT+02:00 pavel@eremina.net eremina.net@gmail.com:
Hi! Check "did=" in your dialog messages? if not present, use match_dialog_mode = 0.
2014-06-10 18:16 GMT+06:00 Gijs Kwakkel kwakkel1000@gmail.com:
client1 <> provider <> kamailio <> asterisk <> client2
when client1 calls client2, everything seems to work. sounds works. after a few seconds the call gets terminated.
after tcpdump syslog and config checks i figured out that asterisk sends 200 OK through kamailio to the provider, after this the provider sends a ACK to kamailio, kamailio however sends this ACK to itself and not to asterisk.
I added the config, syslog and wireshark output (converted)
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Hi again. You can try to use topology hiding in you kamailio( don't forget add some code for message which has uri==myself it present in docs).
I use it and ack processing well.
I think it's because some sip servers can't work with SIP proxy. it created only for PBX.
2014-06-10 18:16 GMT+06:00 Gijs Kwakkel kwakkel1000@gmail.com:
client1 <> provider <> kamailio <> asterisk <> client2
when client1 calls client2, everything seems to work. sounds works. after a few seconds the call gets terminated.
after tcpdump syslog and config checks i figured out that asterisk sends 200 OK through kamailio to the provider, after this the provider sends a ACK to kamailio, kamailio however sends this ACK to itself and not to asterisk.
I added the config, syslog and wireshark output (converted)
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
that seems to work, (i did not add anything new for (has uri==myself)) i'm not sure yet why this is working, so i will study it a bit more. if this doesn't bring up new bugs, then my whole problem seems to be solved.
2014-06-12 9:17 GMT+02:00 pavel@eremina.net eremina.net@gmail.com:
Hi again. You can try to use topology hiding in you kamailio( don't forget add some code for message which has uri==myself it present in docs).
I use it and ack processing well.
I think it's because some sip servers can't work with SIP proxy. it created only for PBX.
2014-06-10 18:16 GMT+06:00 Gijs Kwakkel kwakkel1000@gmail.com:
client1 <> provider <> kamailio <> asterisk <> client2
when client1 calls client2, everything seems to work. sounds works. after a few seconds the call gets terminated.
after tcpdump syslog and config checks i figured out that asterisk sends 200 OK through kamailio to the provider, after this the provider sends a ACK to kamailio, kamailio however sends this ACK to itself and not to asterisk.
I added the config, syslog and wireshark output (converted)
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
so, when you add dialog module your problem gone?
2014-06-12 20:14 GMT+06:00 Gijs Kwakkel kwakkel1000@gmail.com:
that seems to work, (i did not add anything new for (has uri==myself)) i'm not sure yet why this is working, so i will study it a bit more. if this doesn't bring up new bugs, then my whole problem seems to be solved.
2014-06-12 9:17 GMT+02:00 pavel@eremina.net eremina.net@gmail.com:
Hi again.
You can try to use topology hiding in you kamailio( don't forget add some code for message which has uri==myself it present in docs).
I use it and ack processing well.
I think it's because some sip servers can't work with SIP proxy. it created only for PBX.
2014-06-10 18:16 GMT+06:00 Gijs Kwakkel kwakkel1000@gmail.com:
client1 <> provider <> kamailio <> asterisk <> client2
when client1 calls client2, everything seems to work. sounds works. after a few seconds the call gets terminated.
after tcpdump syslog and config checks i figured out that asterisk sends 200 OK through kamailio to the provider, after this the provider sends a ACK to kamailio, kamailio however sends this ACK to itself and not to asterisk.
I added the config, syslog and wireshark output (converted)
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
-
Gijs Kwakkel -
pavel＠eremina.net