Thank you very much for sharing your insights, Barry! I am facing the same problem that Trevor described:
Things are working just fine on their own, but as soon as FreePBX comes into play, calling extensions becomes impossible because of the different tables used. Removing the password from FreePBX (and setting the Kamailio IP in the ACL field) seems to mitigate the issue somewhat, but even though the extension shows as registered in FreePBX, it always shows as busy:
chan_sip.c:23237 handle_response_invite: Failed to authenticate on INVITE to '"xxxxxxxx" sip:xxxxxxxx@198.23.139.21;tag=as72a4117a' -- SIP/1001-00000006 is circuit-busy
I doubt that I can make the necessary modifications even with your hints, but would be willing to pay for your (or anybody's) time solving the matter in a way that (a) leaves the FreePBX installation untouched as far as possible and (b) is easy to apply to subsequent versions. Ideally, it would be a slight modification of Daniel's excellent tutorial at
http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb,
allowing to create extensions in FreePBX. Anybody willing to give it a shot? Just let me know what you would charge and we can talk about it.
On 29 May 2013 10:25, Michael Leuker michael@leuker.me wrote:
Thank you very much for sharing your insights, Barry! I am facing the same problem that Trevor described:
Things are working just fine on their own, but as soon as FreePBX comes into play, calling extensions becomes impossible because of the different tables used. Removing the password from FreePBX (and setting the Kamailio IP in the ACL field) seems to mitigate the issue somewhat, but even though the extension shows as registered in FreePBX, it always shows as busy:
chan_sip.c:23237 handle_response_invite: Failed to authenticate on INVITE to '"xxxxxxxx" sip:xxxxxxxx@198.23.139.21;tag=as72a4117a' -- SIP/1001-00000006 is circuit-busy
Can you do "sip set debug on" on Asterisk and make a call and post the output?
-Barry
Sure, here's the sequence for an inbound call via the "LPhone" trunk that was supposed to go through to extension 1001. The extension was set to "NAT" in the FreePBX settings. Just ask if you need more background.
On Wed, May 29, 2013 at 6:14 PM, Barry Flanagan barry@flanagan.ie wrote:
On 29 May 2013 10:25, Michael Leuker michael@leuker.me wrote:
Thank you very much for sharing your insights, Barry! I am facing the same problem that Trevor described:
Things are working just fine on their own, but as soon as FreePBX comes into play, calling extensions becomes impossible because of the different tables used. Removing the password from FreePBX (and setting the Kamailio IP in the ACL field) seems to mitigate the issue somewhat, but even though the extension shows as registered in FreePBX, it always shows as busy:
chan_sip.c:23237 handle_response_invite: Failed to authenticate on INVITE to '"xxxxxxxx" sip:xxxxxxxx@198.23.139.21;tag=as72a4117a' -- SIP/1001-00000006 is circuit-busy
Can you do "sip set debug on" on Asterisk and make a call and post the output?
-Barry
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
On 29 May 2013 19:23, Michael Leuker michael@leuker.me wrote:
Sure, here's the sequence for an inbound call via the "LPhone" trunk that was supposed to go through to extension 1001. The extension was set to "NAT" in the FreePBX settings. Just ask if you need more background.
The Asterisk part looks fine. It is sending the call to 1001@198.23.139.21:5060 which I presume is your Kamailio instance.
for some reason Kamailio is not recognising the user. Could it be that you do not have the ip 198.23.139.21 set up on Kamailio as an alias, or that the user 1001 is registering to a different domain?
Kamailio would be looking in the location table for "username='1001' AND domain='198.23.139.21'"
You should check the Kamailio logs for what is happening when Asterisk sends it the INVITE for 1001@198.23.139.21
Hope this helps.
-Barry
On Wed, May 29, 2013 at 6:14 PM, Barry Flanagan barry@flanagan.ie wrote:
On 29 May 2013 10:25, Michael Leuker michael@leuker.me wrote:
Thank you very much for sharing your insights, Barry! I am facing the same problem that Trevor described:
Things are working just fine on their own, but as soon as FreePBX comes into play, calling extensions becomes impossible because of the different tables used. Removing the password from FreePBX (and setting the Kamailio IP in the ACL field) seems to mitigate the issue somewhat, but even though the extension shows as registered in FreePBX, it always shows as busy:
chan_sip.c:23237 handle_response_invite: Failed to authenticate on INVITE to '"xxxxxxxx" sip:xxxxxxxx@198.23.139.21;tag=as72a4117a' -- SIP/1001-00000006 is circuit-busy
Can you do "sip set debug on" on Asterisk and make a call and post the output?
-Barry
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
Thank you so much for pointing me in the right direction! It was the missing alias. Now the extensions are working, but there's one more problem: When I call the PBX (echo-test) or from one extension to another, I get a hangup because Asterisk doesn't seem to receive an ACK from the client:
"Retransmission timeout reached on transmission OGNkYTY1ZmJmY2VmMjQ2YmM4MWU1YWY0YjU3NjlhYjA for seqno 2 (Critical Response)"
I have attached the log for two local calls that hangup after about 30s and the sequence for an incoming call from one of the trunks that works without problems. So you can map all the IPs:
-------- 198.23.139.21 (5060): Kamailio 198.23.139.21 (5080): Asterisk -------- 192.168.178.33: Home network local IP 188.105.112.187: Home network external IP -------- 94.75.247.45: Trunk --------
NAT was set to "No - RFC3581" for these captures, but I've tried all other possibilities (including nathelper / rtpproxy) without success. Do you (or anybody else) have any idea where to look in order to solve this problem?
On Thu, May 30, 2013 at 10:36 AM, Barry Flanagan barry@flanagan.ie wrote:
On 29 May 2013 19:23, Michael Leuker michael@leuker.me wrote:
Sure, here's the sequence for an inbound call via the "LPhone" trunk that was supposed to go through to extension 1001. The extension was set to "NAT" in the FreePBX settings. Just ask if you need more background.
The Asterisk part looks fine. It is sending the call to 1001@198.23.139.21:5060 which I presume is your Kamailio instance.
for some reason Kamailio is not recognising the user. Could it be that you do not have the ip 198.23.139.21 set up on Kamailio as an alias, or that the user 1001 is registering to a different domain?
Kamailio would be looking in the location table for "username='1001' AND domain='198.23.139.21'"
You should check the Kamailio logs for what is happening when Asterisk sends it the INVITE for 1001@198.23.139.21
Hope this helps.
-Barry
On Wed, May 29, 2013 at 6:14 PM, Barry Flanagan barry@flanagan.iewrote:
On 29 May 2013 10:25, Michael Leuker michael@leuker.me wrote:
Thank you very much for sharing your insights, Barry! I am facing the same problem that Trevor described:
Things are working just fine on their own, but as soon as FreePBX comes into play, calling extensions becomes impossible because of the different tables used. Removing the password from FreePBX (and setting the Kamailio IP in the ACL field) seems to mitigate the issue somewhat, but even though the extension shows as registered in FreePBX, it always shows as busy:
chan_sip.c:23237 handle_response_invite: Failed to authenticate on INVITE to '"xxxxxxxx" sip:xxxxxxxx@198.23.139.21;tag=as72a4117a' -- SIP/1001-00000006 is circuit-busy
Can you do "sip set debug on" on Asterisk and make a call and post the output?
-Barry
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
On 30 May 2013 21:09, Michael Leuker michael@leuker.me wrote:
Thank you so much for pointing me in the right direction! It was the missing alias. Now the extensions are working, but there's one more problem: When I call the PBX (echo-test) or from one extension to another, I get a hangup because Asterisk doesn't seem to receive an ACK from the client:
"Retransmission timeout reached on transmission OGNkYTY1ZmJmY2VmMjQ2YmM4MWU1YWY0YjU3NjlhYjA for seqno 2 (Critical Response)"
Solution is going to lie in the kamailio logs I think:
"Retransmitting #5 (no NAT) to 198.23.139.21:5060: "
...for whatever reason, Kamailio is not replying to these.
-Barry Flanagan
I have attached the log for two local calls that hangup after about 30s and
the sequence for an incoming call from one of the trunks that works without problems. So you can map all the IPs:
198.23.139.21 (5060): Kamailio 198.23.139.21 (5080): Asterisk
192.168.178.33: Home network local IP 188.105.112.187: Home network external IP
94.75.247.45: Trunk
NAT was set to "No - RFC3581" for these captures, but I've tried all other possibilities (including nathelper / rtpproxy) without success. Do you (or anybody else) have any idea where to look in order to solve this problem?
On Thu, May 30, 2013 at 10:36 AM, Barry Flanagan barry@flanagan.iewrote:
On 29 May 2013 19:23, Michael Leuker michael@leuker.me wrote:
Sure, here's the sequence for an inbound call via the "LPhone" trunk that was supposed to go through to extension 1001. The extension was set to "NAT" in the FreePBX settings. Just ask if you need more background.
The Asterisk part looks fine. It is sending the call to 1001@198.23.139.21:5060 which I presume is your Kamailio instance.
for some reason Kamailio is not recognising the user. Could it be that you do not have the ip 198.23.139.21 set up on Kamailio as an alias, or that the user 1001 is registering to a different domain?
Kamailio would be looking in the location table for "username='1001' AND domain='198.23.139.21'"
You should check the Kamailio logs for what is happening when Asterisk sends it the INVITE for 1001@198.23.139.21
Hope this helps.
-Barry
On Wed, May 29, 2013 at 6:14 PM, Barry Flanagan barry@flanagan.iewrote:
On 29 May 2013 10:25, Michael Leuker michael@leuker.me wrote:
Thank you very much for sharing your insights, Barry! I am facing the same problem that Trevor described:
Things are working just fine on their own, but as soon as FreePBX comes into play, calling extensions becomes impossible because of the different tables used. Removing the password from FreePBX (and setting the Kamailio IP in the ACL field) seems to mitigate the issue somewhat, but even though the extension shows as registered in FreePBX, it always shows as busy:
chan_sip.c:23237 handle_response_invite: Failed to authenticate on INVITE to '"xxxxxxxx" sip:xxxxxxxx@198.23.139.21;tag=as72a4117a' -- SIP/1001-00000006 is circuit-busy
Can you do "sip set debug on" on Asterisk and make a call and post the output?
-Barry
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
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
-
Barry Flanagan -
Michael Leuker