Extension does not ring after some time
Hello everybody, I'll try to explain my strange problem: I have a Freeswitch in the cloud (set following the wiki for amazon ec2) and I cannot bridge calls to extensions (no ring) some seconds after the registration. When register the ip phone is all ok but after a while this is what i can see on the console if i try to call his extension: http://pastebin.com/YT4Pgi7K i cannot hear any ring from leg a and the phone on the leg b doens't ring and after some seconds the call is hanged up with cause "[CS_CONSUME_MEDIA] [NO_ANSWER]" obtaining this on the fs console. 2014-04-17 23:02:54.013897 [DEBUG] switch_channel.c:3187 Send signal sofia/internal/900 [KILL] 2014-04-17 23:02:54.013897 [DEBUG] switch_core_session.c:1351 Send signal sofia/internal/900 [BREAK] 2014-04-17 23:02:54.013897 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/900) Running State Change CS_HANGUP 2014-04-17 23:02:54.013897 [DEBUG] switch_core_state_machine.c:678 (sofia/internal/900) Callstate Change DOWN -> HANGUP 2014-04-17 23:02:54.013897 [DEBUG] switch_core_state_machine.c:680 (sofia/internal/900) State HANGUP 2014-04-17 23:02:54.013897 [DEBUG] mod_sofia.c:506 Channel sofia/internal/900 hanging up, cause: NO_ANSWER 2014-04-17 23:02:54.013897 [INFO] mod_dptools.c:3201 Originate Failed. Cause: NO_ANSWER If i re-register the phone i can receive call for some seconds and than the problem returns. How can i manage that? Thanks in advice crazygabry
Mr.Gab <crazygabry@...> writes:
Hello everybody, I'll try to explain my strange problem: I have a Freeswitch in the cloud
(set following the wiki for amazon ec2) and I cannot bridge calls to extensions (no ring) some seconds after the registration. When register the ip phone is all ok but after a while this is what i can see on the console if i try to call his extension:http://pastebin.com/YT4Pgi7K
i cannot hear any ring from leg a and the phone on the leg b doens't ring
and after some seconds the call is hanged up with cause "[CS_CONSUME_MEDIA] [NO_ANSWER]" obtaining this on the fs console.
2014-04-17 23:02:54.013897 [DEBUG] switch_channel.c:3187 Send signal
sofia/internal/900 [KILL]
2014-04-17 23:02:54.013897 [DEBUG] switch_core_session.c:1351 Send signal sofia/internal/900 [BREAK]
2014-04-17 23:02:54.013897 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/900) Running State Change CS_HANGUP 2014-04-17 23:02:54.013897 [DEBUG] switch_core_state_machine.c:678 (sofia/internal/900) Callstate Change DOWN -> HANGUP
2014-04-17 23:02:54.013897 [DEBUG] switch_core_state_machine.c:680 (sofia/internal/900) State HANGUP 2014-04-17 23:02:54.013897 [DEBUG] mod_sofia.c:506 Channel sofia/internal/900 hanging up, cause: NO_ANSWER
2014-04-17 23:02:54.013897 [INFO] mod_dptools.c:3201 Originate Failed. Cause: NO_ANSWER
If i re-register the phone i can receive call for some seconds and than the problem returns.
How can i manage that?
Thanks in advice crazygabry
Implementing sofia_contact on bridge function I discovered one thing that could help debugging described in the previous message. If i try to call the extension few seconds after the registration sofia_contact returns "sofia/internal/sip:900@X.X.X.X:PORT", with X.X.X.X public ip address of the phone, but after some minutes sofia_contact returns "sofia/internal/sip:900@Y.Y.Y.Y:PORT" with Y.Y.Y.Y local ip of the phone. I don't undestand if is a FS problem or client problem: I detected this problem with a snom 300 and a thomson ST2030, but I didn't detected the problem with x-lite softphone and siemens gigaset a510IP. still thanks in andvise :) crazygabry
Clear firewall issue, have allowed the default ports? I notice the wiki doesn't mention port 5080 https://wiki.freeswitch.org/wiki/Amazon_ec2 /b Sent from my iPhone
On Apr 17, 2014, at 6:33 PM, Mr.Gab <crazygabry@gmail.com> wrote:
Mr.Gab <crazygabry@...> writes:
Hello everybody, I'll try to explain my strange problem: I have a Freeswitch in the cloud
(set following the wiki for amazon ec2) and I cannot bridge calls to extensions (no ring) some seconds after the registration. When register the ip phone is all ok but after a while this is what i can see on the console if i try to call his extension:http://pastebin.com/YT4Pgi7K
i cannot hear any ring from leg a and the phone on the leg b doens't ring
and after some seconds the call is hanged up with cause "[CS_CONSUME_MEDIA] [NO_ANSWER]" obtaining this on the fs console.
2014-04-17 23:02:54.013897 [DEBUG] switch_channel.c:3187 Send signal
sofia/internal/900 [KILL]
2014-04-17 23:02:54.013897 [DEBUG] switch_core_session.c:1351 Send signal sofia/internal/900 [BREAK]
2014-04-17 23:02:54.013897 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/900) Running State Change CS_HANGUP 2014-04-17 23:02:54.013897 [DEBUG] switch_core_state_machine.c:678 (sofia/internal/900) Callstate Change DOWN -> HANGUP
2014-04-17 23:02:54.013897 [DEBUG] switch_core_state_machine.c:680 (sofia/internal/900) State HANGUP 2014-04-17 23:02:54.013897 [DEBUG] mod_sofia.c:506 Channel sofia/internal/900 hanging up, cause: NO_ANSWER
2014-04-17 23:02:54.013897 [INFO] mod_dptools.c:3201 Originate Failed. Cause: NO_ANSWER
If i re-register the phone i can receive call for some seconds and than the problem returns.
How can i manage that?
Thanks in advice crazygabry
Implementing sofia_contact on bridge function I discovered one thing that could help debugging described in the previous message. If i try to call the extension few seconds after the registration sofia_contact returns "sofia/internal/sip:900@X.X.X.X:PORT", with X.X.X.X public ip address of the phone, but after some minutes sofia_contact returns "sofia/internal/sip:900@Y.Y.Y.Y:PORT" with Y.Y.Y.Y local ip of the phone. I don't undestand if is a FS problem or client problem: I detected this problem with a snom 300 and a thomson ST2030, but I didn't detected the problem with x-lite softphone and siemens gigaset a510IP.
still thanks in andvise :) crazygabry
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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Brian West <brian@...> writes:
Clear firewall issue, have allowed the default ports?
I notice the wiki doesn't mention port 5080
https://wiki.freeswitch.org/wiki/Amazon_ec2
/b
Sent from my iPhone
Thank you for the reply, but i still have the problem :( I noticed another strange thing that would make me think about a registration issue. When i detect the problem if look to the registration list by submitting "sofia status profile internal reg" i see that there are multiple registration of the same phone (all it's the same except CALL-ID). For example, now i can see only one registration for my phone and it seems to work fine. Thank for your help crazygabry
Have you tried to turn on NAT Keep Alive on the client? -----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Mr.Gab Sent: Friday, April 18, 2014 4:26 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Extension does not ring after some time Brian West <brian@...> writes:
Clear firewall issue, have allowed the default ports?
I notice the wiki doesn't mention port 5080
https://wiki.freeswitch.org/wiki/Amazon_ec2
/b
Sent from my iPhone
Thank you for the reply, but i still have the problem :( I noticed another strange thing that would make me think about a registration issue. When i detect the problem if look to the registration list by submitting "sofia status profile internal reg" i see that there are multiple registration of the same phone (all it's the same except CALL-ID). For example, now i can see only one registration for my phone and it seems to work fine. Thank for your help crazygabry _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
do you think is a client problem or server
NAT routers need to keep a list of UDP connections open to know the internal/external ports to map between and what connections are open to allow the replies through the firewall. Since there's no explicit disconnect in UDP it relies exclusively on timeouts to remove old connections from the table when no longer in use, to prevent the mapping table becoming full. Once it's removed from the table the router is no longer going to know the internal client to send the INVITE from the server to, which could be why you stop receiving the call on the client. From the server's perspective it probably times out. Either side sending packets (eg keepalives) will keep the connection 'recently active' and therefore the mapping open. Check the SIP trace on FreeSWITCH (sofia global siptrace on). See if the phone gives a response to the INVITE. If the phone gives a reply it's getting through the NAT and so that's not the issue. If you get no reply it's highly likely it is the NAT router. can you suggest me a value? The best value is going to depend on the router - on what timeout value it has, how large the table is, and how many connections you have open through it. 30s is probably a good setting to go. You can tweak it up/down from there if need be. On 18 April 2014 14:48, Mr.Gab <crazygabry@gmail.com> wrote:
Sean Devoy <sdevoy@...> writes:
Have you tried to turn on NAT Keep Alive on the client?
can you suggest me a value? :) do you think is a client problem or server problem?
Thank you crazygabry
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Steven Ayre <steveayre@...> writes:
do you think is a client problem or server
NAT routers need to keep a list of UDP connections open to know the
internal/external ports to map between and what connections are open to allow the replies through the firewall. Since there's no explicit disconnect in UDP it relies exclusively on timeouts to remove old connections from the table when no longer in use, to prevent the mapping table becoming full.
Once it's removed from the table the router is no longer going to know the
internal client to send the INVITE from the server to, which could be why you stop receiving the call on the client. From the server's perspective it probably times out.
Either side sending packets (eg keepalives) will keep the connection
'recently active' and therefore the mapping open.
Check the SIP trace on FreeSWITCH (sofia global siptrace on). See if the
phone gives a response to the INVITE. If the phone gives a reply it's getting through the NAT and so that's not the issue. If you get no reply it's highly likely it is the NAT router.
can you suggest me a value?
The best value is going to depend on the router - on what timeout value it
has, how large the table is, and how many connections you have open through it.
30s is probably a good setting to go. You can tweak it up/down from there
if need be. thank you very much! unfortunately now i can't try this settings on the phones. i'll keep you informed asap :) Happy Easter!
participants (4)
-
Brian West
-
Mr.Gab
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Sean Devoy
-
Steven Ayre