[Freeswitch-users] Keep alive for SIP trunk between Asterisk and Freeswitch

Discussion:

Daniel Neubert

2010-07-28 07:25:15 UTC

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 Gr??en,
Daniel

Steven Ayre

2010-07-28 08:08:16 UTC

Permalink

Where & why does the call fail?

Do you have any log file output?

-Steve

Post by Daniel Neubert
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
<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 Gr??en,
Daniel
_______________________________________________
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FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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Daniel Neubert

2010-07-28 09:33:59 UTC

Permalink

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 Gr??en,
Daniel

Post by Steven Ayre
Where & why does the call fail?
Do you have any log file output?
-Steve
On 28 July 2010 08:25, Daniel Neubert <daniel.neubert at solomo.de
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
<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 Gr??en,
Daniel
_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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David Ponzone

2010-07-28 13:42:45 UTC

Permalink

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 at 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 pi?ces jointes sont confidentiels et ?tablis
? l'intention exclusive de ses destinataires. Toute utilisation ou
diffusion non autoris?e est interdite. Tout message ?lectronique est
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au
titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si
vous n'?tes pas destinataire de ce message, merci de le d?truire
imm?diatement et d'avertir l'exp?diteur.

Post by Daniel Neubert
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 Gr??en,
Daniel

Post by Steven Ayre
Where & why does the call fail?
Do you have any log file output?
-Steve
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
<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 Gr??en,
Daniel
_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at 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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FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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Daniel Neubert

2010-07-28 13:52:32 UTC

Permalink

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 at voip-test) Callstate Change EARLY -> HANGUP
2010-07-28 15:49:26.644905 [NOTICE] mod_dptools.c:2456 Hangup
sofia/internal/test01 at 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 Gr??en,
Daniel

Post by Daniel Neubert
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 Gr??en,
Daniel

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Anthony Minessale

2010-07-28 19:17:06 UTC

Permalink

you could just omit the ping param if asterisk can't operate properly with
that feature.

Post by Daniel Neubert
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 at voip-test) Callstate Change EARLY -> HANGUP
2010-07-28 15:49:26.644905 [NOTICE] mod_dptools.c:2456 Hangup
sofia/internal/test01 at 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 Gr??en,
Daniel
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 Gr??en,
Daniel
Where & why does the call fail?
Do you have any log file output?
-Steve

--
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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Nyamul Hassan

2010-07-28 21:12:00 UTC

Permalink

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 <

Post by Anthony Minessale
you could just omit the ping param if asterisk can't operate properly with
that feature.

Post by Daniel Neubert
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 at voip-test) Callstate Change EARLY -> HANGUP
2010-07-28 15:49:26.644905 [NOTICE] mod_dptools.c:2456 Hangup
sofia/internal/test01 at 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 Gr??en,
Daniel
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 Gr??en,
Daniel
Where & why does the call fail?
Do you have any log file output?
-Steve

--
Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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Daniel Neubert

2010-07-29 08:06:38 UTC

Permalink

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 Gr??en,
Daniel

Post by Anthony Minessale
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
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
[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 at voip-test) Callstate Change EARLY -> HANGUP
2010-07-28 15:49:26.644905 [NOTICE] mod_dptools.c:2456 Hangup
sofia/internal/test01 at 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 Gr??en,
Daniel

Post by Daniel Neubert
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 Gr??en,
Daniel

Post by Steven Ayre
Where & why does the call fail?
Do you have any log file output?
-Steve
On 28 July 2010 08:25, Daniel Neubert <daniel.neubert at solomo.de
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
<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 Gr??en,
Daniel
_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
<mailto:FreeSWITCH-users at 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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http://www.freeswitch.org
--
Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm
MSN:anthony_minessale at hotmail.com
<mailto:MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
<mailto:PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
<mailto:sip%3A888 at conference.freeswitch.org>
googletalk:conf+888 at conference.freeswitch.org
<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900

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