10 Jun
2014
10 Jun
'14
3:16 p.m.
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)
10 Jun
10 Jun
6:46 p.m.
New subject: inbound kamailio dispatcher faulty routing of ack after 200 OK
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(a)gmail.com>om>:
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)
_______________________________________________
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11 Jun
11 Jun
11:32 a.m.
New subject: inbound kamailio dispatcher faulty routing of ack after 200 OK
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(a)eremina.net <eremina.net(a)gmail.com>om>:
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(a)gmail.com>om>:
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(a)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(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
13 Jun
13 Jun
6:31 a.m.
New subject: inbound kamailio dispatcher faulty routing of ack after 200 OK
Can you show here this errors?
12.06.2014 13:37 пользователь "Gijs Kwakkel" <kwakkel1000(a)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(a)eremina.net <eremina.net(a)gmail.com>om>:
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(a)gmail.com>om>:
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
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(a)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(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
12 Jun
12 Jun
10:17 a.m.
New subject: inbound kamailio dispatcher faulty routing of ack after 200 OK
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(a)gmail.com>om>:
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(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
5:14 p.m.
New subject: inbound kamailio dispatcher faulty routing of ack after 200 OK
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(a)eremina.net <eremina.net(a)gmail.com>om>:
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(a)gmail.com>om>:
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(a)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(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
13 Jun
13 Jun
8:31 p.m.
New subject: inbound kamailio dispatcher faulty routing of ack after 200 OK
so, when you add dialog module your problem gone?
2014-06-12 20:14 GMT+06:00 Gijs Kwakkel <kwakkel1000(a)gmail.com>om>:
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(a)eremina.net <eremina.net(a)gmail.com>om>:
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(a)gmail.com>om>:
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
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(a)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(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
3819
days inactive
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6 comments
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participants (2)
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Gijs Kwakkel -
pavel＠eremina.net