Keep alive for SIP trunk between Asterisk and Freeswitch
Hi, we've set up a SIP trunk between Asterisk (used as MediaGateway to SS7-Network for PSTN access) and Freeswitch. Everything works fine except one "little" issue: If there have been no calls using the SIP trunk it becomes unuseable from Freeswitch side. PSTN <- SS7/ISUP -> Asterisk <- SIP Trunk -> Freeswitch <- SIP/RTP -> VoIP Clients If a user tries to originate the call that is routed to one of our MediaGateways while SIP trunk is "stale", the call will fail. The trunk can be made available again by calling in via PSTN -> Asterisk -> SIP-Trunk This is our gateway configuration (tried using low values for expire-seconds, ping and retry-seconds to keep the connection up: <gateway name="voip-int-test"> <param name="username" value="voip-ext-test"/> <param name="password" value="freeswitch"/> <param name="proxy" value="172.31.45.43"/> <param name="register" value="false"/> <param name="expire-seconds" value="15"/> <param name="ping" value="5"/> <param name="retry-seconds" value="5"/> <param name="context" value="default"/> <param name="apply-inbound-acl" value="voip-int-test"/> <param name="caller-id-in-from" value="true"/> </gateway> -- Best regards / Mit freundlichen Grüßen, Daniel
Where & why does the call fail? Do you have any log file output? -Steve On 28 July 2010 08:25, Daniel Neubert <daniel.neubert@solomo.de> wrote:
Hi,
we've set up a SIP trunk between Asterisk (used as MediaGateway to SS7-Network for PSTN access) and Freeswitch.
Everything works fine except one "little" issue: If there have been no calls using the SIP trunk it becomes unuseable from Freeswitch side.
PSTN <- SS7/ISUP -> Asterisk <- SIP Trunk -> Freeswitch <- SIP/RTP -> VoIP Clients
If a user tries to originate the call that is routed to one of our MediaGateways while SIP trunk is "stale", the call will fail. The trunk can be made available again by calling in via PSTN -> Asterisk -> SIP-Trunk
This is our gateway configuration (tried using low values for expire-seconds, ping and retry-seconds to keep the connection up:
<gateway name="voip-int-test"> <param name="username" value="voip-ext-test"/> <param name="password" value="freeswitch"/> <param name="proxy" value="172.31.45.43"/> <param name="register" value="false"/> <param name="expire-seconds" value="15"/> <param name="ping" value="5"/> <param name="retry-seconds" value="5"/> <param name="context" value="default"/> <param name="apply-inbound-acl" value="voip-int-test"/> <param name="caller-id-in-from" value="true"/> </gateway>
--
Best regards / Mit freundlichen Grüßen, Daniel
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The call fails because the desired gateway is down. Logs are not available at the moment and issue cannot be reproduced on demand. I'll take logs as soon as this occurs again. Best regards / Mit freundlichen Grüßen, Daniel On 28.07.2010 10:08, Steven Ayre wrote:
Where & why does the call fail?
Do you have any log file output?
-Steve
On 28 July 2010 08:25, Daniel Neubert <daniel.neubert@solomo.de <mailto:daniel.neubert@solomo.de>> wrote:
Hi,
we've set up a SIP trunk between Asterisk (used as MediaGateway to SS7-Network for PSTN access) and Freeswitch.
Everything works fine except one "little" issue: If there have been no calls using the SIP trunk it becomes unuseable from Freeswitch side.
PSTN <- SS7/ISUP -> Asterisk <- SIP Trunk -> Freeswitch <- SIP/RTP -> VoIP Clients
If a user tries to originate the call that is routed to one of our MediaGateways while SIP trunk is "stale", the call will fail. The trunk can be made available again by calling in via PSTN -> Asterisk -> SIP-Trunk
This is our gateway configuration (tried using low values for expire-seconds, ping and retry-seconds to keep the connection up:
<gateway name="voip-int-test"> <param name="username" value="voip-ext-test"/> <param name="password" value="freeswitch"/> <param name="proxy" value="172.31.45.43"/> <param name="register" value="false"/> <param name="expire-seconds" value="15"/> <param name="ping" value="5"/> <param name="retry-seconds" value="5"/> <param name="context" value="default"/> <param name="apply-inbound-acl" value="voip-int-test"/> <param name="caller-id-in-from" value="true"/> </gateway>
--
Best regards / Mit freundlichen Grüßen, Daniel
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Do you see the SIP OPTIONS sent by FS to Asterisk every 5 seconds ? Do you see Asterisk's replies ? What is it ? (depending on what Asterisk replies as SIP Error code, FS could decide to down the gateway) Normally, Asterisk should reply 404 or 200. David Ponzone Direction Technique email: david.ponzone@ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur. Le 28/07/2010 à 11:33, Daniel Neubert a écrit :
The call fails because the desired gateway is down.
Logs are not available at the moment and issue cannot be reproduced on demand. I'll take logs as soon as this occurs again. Best regards / Mit freundlichen Grüßen, Daniel On 28.07.2010 10:08, Steven Ayre wrote:
Where & why does the call fail?
Do you have any log file output?
-Steve
On 28 July 2010 08:25, Daniel Neubert <daniel.neubert@solomo.de> wrote: Hi,
we've set up a SIP trunk between Asterisk (used as MediaGateway to SS7-Network for PSTN access) and Freeswitch.
Everything works fine except one "little" issue: If there have been no calls using the SIP trunk it becomes unuseable from Freeswitch side.
PSTN <- SS7/ISUP -> Asterisk <- SIP Trunk -> Freeswitch <- SIP/RTP -> VoIP Clients
If a user tries to originate the call that is routed to one of our MediaGateways while SIP trunk is "stale", the call will fail. The trunk can be made available again by calling in via PSTN -> Asterisk -> SIP-Trunk
This is our gateway configuration (tried using low values for expire-seconds, ping and retry-seconds to keep the connection up:
<gateway name="voip-int-test"> <param name="username" value="voip-ext-test"/> <param name="password" value="freeswitch"/> <param name="proxy" value="172.31.45.43"/> <param name="register" value="false"/> <param name="expire-seconds" value="15"/> <param name="ping" value="5"/> <param name="retry-seconds" value="5"/> <param name="context" value="default"/> <param name="apply-inbound-acl" value="voip-int-test"/> <param name="caller-id-in-from" value="true"/> </gateway>
--
Best regards / Mit freundlichen Grüßen, Daniel
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Now I have a trace from Freeswitch log: 2010-07-28 15:49:26.630733 [ERR] mod_sofia.c:3674 Gateway is down! 2010-07-28 15:49:26.630733 [NOTICE] mod_sofia.c:3984 Close Channel N/A [CS_NEW] 2010-07-28 15:49:26.640905 [DEBUG] switch_core_state_machine.c:430 () Running State Change CS_DESTROY 2010-07-28 15:49:26.640905 [DEBUG] switch_core_state_machine.c:440 (N/A) State DESTROY 2010-07-28 15:49:26.640905 [DEBUG] mod_sofia.c:358 N/A SOFIA DESTROY 2010-07-28 15:49:26.640905 [DEBUG] switch_core_state_machine.c:440 (N/A) State DESTROY going to sleep 2010-07-28 15:49:26.640905 [ERR] switch_ivr_originate.c:2623 Cannot create outgoing channel of type [sofia] cause: [NETWORK_OUT_OF_ORDER] 2010-07-28 15:49:26.642906 [DEBUG] switch_ivr_originate.c:3431 Originate Resulted in Error Cause: 38 [NETWORK_OUT_OF_ORDER] 2010-07-28 15:49:26.642906 [INFO] mod_dptools.c:2393 Originate Failed. Cause: NETWORK_OUT_OF_ORDER 2010-07-28 15:49:26.644905 [DEBUG] switch_channel.c:2309 (sofia/internal/test01@voip-test) Callstate Change EARLY -> HANGUP 2010-07-28 15:49:26.644905 [NOTICE] mod_dptools.c:2456 Hangup sofia/internal/test01@voip-test [CS_EXECUTE] [NETWORK_OUT_OF_ORDER] Directly after that I called in from PSTN -> SS7 -> Asterisk so the gateway came up again and outbound call was possible. Best regards / Mit freundlichen Grüßen, Daniel On 28.07.2010 11:33, Daniel Neubert wrote:
The call fails because the desired gateway is down.
Logs are not available at the moment and issue cannot be reproduced on demand. I'll take logs as soon as this occurs again. Best regards / Mit freundlichen Grüßen, Daniel
On 28.07.2010 10:08, Steven Ayre wrote:
Where & why does the call fail?
Do you have any log file output?
-Steve
On 28 July 2010 08:25, Daniel Neubert <daniel.neubert@solomo.de <mailto:daniel.neubert@solomo.de>> wrote:
Hi,
we've set up a SIP trunk between Asterisk (used as MediaGateway to SS7-Network for PSTN access) and Freeswitch.
Everything works fine except one "little" issue: If there have been no calls using the SIP trunk it becomes unuseable from Freeswitch side.
PSTN <- SS7/ISUP -> Asterisk <- SIP Trunk -> Freeswitch <- SIP/RTP -> VoIP Clients
If a user tries to originate the call that is routed to one of our MediaGateways while SIP trunk is "stale", the call will fail. The trunk can be made available again by calling in via PSTN -> Asterisk -> SIP-Trunk
This is our gateway configuration (tried using low values for expire-seconds, ping and retry-seconds to keep the connection up:
<gateway name="voip-int-test"> <param name="username" value="voip-ext-test"/> <param name="password" value="freeswitch"/> <param name="proxy" value="172.31.45.43"/> <param name="register" value="false"/> <param name="expire-seconds" value="15"/> <param name="ping" value="5"/> <param name="retry-seconds" value="5"/> <param name="context" value="default"/> <param name="apply-inbound-acl" value="voip-int-test"/> <param name="caller-id-in-from" value="true"/> </gateway>
--
Best regards / Mit freundlichen Grüßen, Daniel
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you could just omit the ping param if asterisk can't operate properly with that feature. On Wed, Jul 28, 2010 at 8:52 AM, Daniel Neubert <daniel.neubert@solomo.de>wrote:
Now I have a trace from Freeswitch log:
2010-07-28 15:49:26.630733 [ERR] mod_sofia.c:3674 Gateway is down! 2010-07-28 15:49:26.630733 [NOTICE] mod_sofia.c:3984 Close Channel N/A [CS_NEW] 2010-07-28 15:49:26.640905 [DEBUG] switch_core_state_machine.c:430 () Running State Change CS_DESTROY 2010-07-28 15:49:26.640905 [DEBUG] switch_core_state_machine.c:440 (N/A) State DESTROY 2010-07-28 15:49:26.640905 [DEBUG] mod_sofia.c:358 N/A SOFIA DESTROY 2010-07-28 15:49:26.640905 [DEBUG] switch_core_state_machine.c:440 (N/A) State DESTROY going to sleep 2010-07-28 15:49:26.640905 [ERR] switch_ivr_originate.c:2623 Cannot create outgoing channel of type [sofia] cause: [NETWORK_OUT_OF_ORDER] 2010-07-28 15:49:26.642906 [DEBUG] switch_ivr_originate.c:3431 Originate Resulted in Error Cause: 38 [NETWORK_OUT_OF_ORDER] 2010-07-28 15:49:26.642906 [INFO] mod_dptools.c:2393 Originate Failed. Cause: NETWORK_OUT_OF_ORDER 2010-07-28 15:49:26.644905 [DEBUG] switch_channel.c:2309 (sofia/internal/test01@voip-test) Callstate Change EARLY -> HANGUP 2010-07-28 15:49:26.644905 [NOTICE] mod_dptools.c:2456 Hangup sofia/internal/test01@voip-test [CS_EXECUTE] [NETWORK_OUT_OF_ORDER]
Directly after that I called in from PSTN -> SS7 -> Asterisk so the gateway came up again and outbound call was possible.
Best regards / Mit freundlichen Grüßen, Daniel
On 28.07.2010 11:33, Daniel Neubert wrote:
The call fails because the desired gateway is down.
Logs are not available at the moment and issue cannot be reproduced on demand. I'll take logs as soon as this occurs again.
Best regards / Mit freundlichen Grüßen, Daniel
On 28.07.2010 10:08, Steven Ayre wrote:
Where & why does the call fail?
Do you have any log file output?
-Steve
On 28 July 2010 08:25, Daniel Neubert <daniel.neubert@solomo.de> wrote:
Hi,
we've set up a SIP trunk between Asterisk (used as MediaGateway to SS7-Network for PSTN access) and Freeswitch.
Everything works fine except one "little" issue: If there have been no calls using the SIP trunk it becomes unuseable from Freeswitch side.
PSTN <- SS7/ISUP -> Asterisk <- SIP Trunk -> Freeswitch <- SIP/RTP -> VoIP Clients
If a user tries to originate the call that is routed to one of our MediaGateways while SIP trunk is "stale", the call will fail. The trunk can be made available again by calling in via PSTN -> Asterisk -> SIP-Trunk
This is our gateway configuration (tried using low values for expire-seconds, ping and retry-seconds to keep the connection up:
<gateway name="voip-int-test"> <param name="username" value="voip-ext-test"/> <param name="password" value="freeswitch"/> <param name="proxy" value="172.31.45.43"/> <param name="register" value="false"/> <param name="expire-seconds" value="15"/> <param name="ping" value="5"/> <param name="retry-seconds" value="5"/> <param name="context" value="default"/> <param name="apply-inbound-acl" value="voip-int-test"/> <param name="caller-id-in-from" value="true"/> </gateway>
--
Best regards / Mit freundlichen Grüßen, Daniel
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-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:+19193869900
If you have full control and access to both the FS and Asterisk, you could "not register" either of them to the other at all. You could just add FS IP as a "peer" in Asterisk. And, in FS, you could just allow calls from the Asterisk based on IP. Assuming, of course, no NAT is involved between FS and Asterisk. Regards HASSAN On Thu, Jul 29, 2010 at 01:17, Anthony Minessale < anthony.minessale@gmail.com> wrote:
you could just omit the ping param if asterisk can't operate properly with that feature.
On Wed, Jul 28, 2010 at 8:52 AM, Daniel Neubert <daniel.neubert@solomo.de>wrote:
Now I have a trace from Freeswitch log:
2010-07-28 15:49:26.630733 [ERR] mod_sofia.c:3674 Gateway is down! 2010-07-28 15:49:26.630733 [NOTICE] mod_sofia.c:3984 Close Channel N/A [CS_NEW] 2010-07-28 15:49:26.640905 [DEBUG] switch_core_state_machine.c:430 () Running State Change CS_DESTROY 2010-07-28 15:49:26.640905 [DEBUG] switch_core_state_machine.c:440 (N/A) State DESTROY 2010-07-28 15:49:26.640905 [DEBUG] mod_sofia.c:358 N/A SOFIA DESTROY 2010-07-28 15:49:26.640905 [DEBUG] switch_core_state_machine.c:440 (N/A) State DESTROY going to sleep 2010-07-28 15:49:26.640905 [ERR] switch_ivr_originate.c:2623 Cannot create outgoing channel of type [sofia] cause: [NETWORK_OUT_OF_ORDER] 2010-07-28 15:49:26.642906 [DEBUG] switch_ivr_originate.c:3431 Originate Resulted in Error Cause: 38 [NETWORK_OUT_OF_ORDER] 2010-07-28 15:49:26.642906 [INFO] mod_dptools.c:2393 Originate Failed. Cause: NETWORK_OUT_OF_ORDER 2010-07-28 15:49:26.644905 [DEBUG] switch_channel.c:2309 (sofia/internal/test01@voip-test) Callstate Change EARLY -> HANGUP 2010-07-28 15:49:26.644905 [NOTICE] mod_dptools.c:2456 Hangup sofia/internal/test01@voip-test [CS_EXECUTE] [NETWORK_OUT_OF_ORDER]
Directly after that I called in from PSTN -> SS7 -> Asterisk so the gateway came up again and outbound call was possible.
Best regards / Mit freundlichen Grüßen, Daniel
On 28.07.2010 11:33, Daniel Neubert wrote:
The call fails because the desired gateway is down.
Logs are not available at the moment and issue cannot be reproduced on demand. I'll take logs as soon as this occurs again.
Best regards / Mit freundlichen Grüßen, Daniel
On 28.07.2010 10:08, Steven Ayre wrote:
Where & why does the call fail?
Do you have any log file output?
-Steve
On 28 July 2010 08:25, Daniel Neubert <daniel.neubert@solomo.de> wrote:
Hi,
we've set up a SIP trunk between Asterisk (used as MediaGateway to SS7-Network for PSTN access) and Freeswitch.
Everything works fine except one "little" issue: If there have been no calls using the SIP trunk it becomes unuseable from Freeswitch side.
PSTN <- SS7/ISUP -> Asterisk <- SIP Trunk -> Freeswitch <- SIP/RTP -> VoIP Clients
If a user tries to originate the call that is routed to one of our MediaGateways while SIP trunk is "stale", the call will fail. The trunk can be made available again by calling in via PSTN -> Asterisk -> SIP-Trunk
This is our gateway configuration (tried using low values for expire-seconds, ping and retry-seconds to keep the connection up:
<gateway name="voip-int-test"> <param name="username" value="voip-ext-test"/> <param name="password" value="freeswitch"/> <param name="proxy" value="172.31.45.43"/> <param name="register" value="false"/> <param name="expire-seconds" value="15"/> <param name="ping" value="5"/> <param name="retry-seconds" value="5"/> <param name="context" value="default"/> <param name="apply-inbound-acl" value="voip-int-test"/> <param name="caller-id-in-from" value="true"/> </gateway>
--
Best regards / Mit freundlichen Grüßen, Daniel
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-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:+19193869900
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I think solution to this issue might be to configure qualify=yes in sip.conf in Asterisk. I'll monitor this solution for a few days but it looks good for now. I'll try to dismiss the ping feature but I like to have it's events via ESL interface... Best regards / Mit freundlichen Grüßen, Daniel On 28.07.2010 21:17, Anthony Minessale wrote:
you could just omit the ping param if asterisk can't operate properly with that feature.
On Wed, Jul 28, 2010 at 8:52 AM, Daniel Neubert <daniel.neubert@solomo.de <mailto:daniel.neubert@solomo.de>> wrote:
Now I have a trace from Freeswitch log:
2010-07-28 15:49:26.630733 [ERR] mod_sofia.c:3674 Gateway is down! 2010-07-28 15:49:26.630733 [NOTICE] mod_sofia.c:3984 Close Channel N/A [CS_NEW] 2010-07-28 15:49:26.640905 [DEBUG] switch_core_state_machine.c:430 () Running State Change CS_DESTROY 2010-07-28 15:49:26.640905 [DEBUG] switch_core_state_machine.c:440 (N/A) State DESTROY 2010-07-28 15:49:26.640905 [DEBUG] mod_sofia.c:358 N/A SOFIA DESTROY 2010-07-28 15:49:26.640905 [DEBUG] switch_core_state_machine.c:440 (N/A) State DESTROY going to sleep 2010-07-28 15:49:26.640905 [ERR] switch_ivr_originate.c:2623 Cannot create outgoing channel of type [sofia] cause: [NETWORK_OUT_OF_ORDER] 2010-07-28 15:49:26.642906 [DEBUG] switch_ivr_originate.c:3431 Originate Resulted in Error Cause: 38 [NETWORK_OUT_OF_ORDER] 2010-07-28 15:49:26.642906 [INFO] mod_dptools.c:2393 Originate Failed. Cause: NETWORK_OUT_OF_ORDER 2010-07-28 15:49:26.644905 [DEBUG] switch_channel.c:2309 (sofia/internal/test01@voip-test) Callstate Change EARLY -> HANGUP 2010-07-28 15:49:26.644905 [NOTICE] mod_dptools.c:2456 Hangup sofia/internal/test01@voip-test [CS_EXECUTE] [NETWORK_OUT_OF_ORDER]
Directly after that I called in from PSTN -> SS7 -> Asterisk so the gateway came up again and outbound call was possible.
Best regards / Mit freundlichen Grüßen, Daniel
On 28.07.2010 11:33, Daniel Neubert wrote:
The call fails because the desired gateway is down.
Logs are not available at the moment and issue cannot be reproduced on demand. I'll take logs as soon as this occurs again. Best regards / Mit freundlichen Grüßen, Daniel
On 28.07.2010 10:08, Steven Ayre wrote:
Where & why does the call fail?
Do you have any log file output?
-Steve
On 28 July 2010 08:25, Daniel Neubert <daniel.neubert@solomo.de <mailto:daniel.neubert@solomo.de>> wrote:
Hi,
we've set up a SIP trunk between Asterisk (used as MediaGateway to SS7-Network for PSTN access) and Freeswitch.
Everything works fine except one "little" issue: If there have been no calls using the SIP trunk it becomes unuseable from Freeswitch side.
PSTN <- SS7/ISUP -> Asterisk <- SIP Trunk -> Freeswitch <- SIP/RTP -> VoIP Clients
If a user tries to originate the call that is routed to one of our MediaGateways while SIP trunk is "stale", the call will fail. The trunk can be made available again by calling in via PSTN -> Asterisk -> SIP-Trunk
This is our gateway configuration (tried using low values for expire-seconds, ping and retry-seconds to keep the connection up:
<gateway name="voip-int-test"> <param name="username" value="voip-ext-test"/> <param name="password" value="freeswitch"/> <param name="proxy" value="172.31.45.43"/> <param name="register" value="false"/> <param name="expire-seconds" value="15"/> <param name="ping" value="5"/> <param name="retry-seconds" value="5"/> <param name="context" value="default"/> <param name="apply-inbound-acl" value="voip-int-test"/> <param name="caller-id-in-from" value="true"/> </gateway>
--
Best regards / Mit freundlichen Grüßen, Daniel
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org <mailto: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
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-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com <mailto:MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com <mailto:PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
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participants (5)
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Anthony Minessale
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Daniel Neubert
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David Ponzone
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Nyamul Hassan
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Steven Ayre