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

[Nyamul Hassan]

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 at 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 at 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 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 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 at 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 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
>>>
>>
>>
>> _______________________________________________
>> 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
>>
>>
>
>
> --
> 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
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100729/58665e4d/attachment.html