[Freeswitch-users] NO_ROUTE_DESTINATION

[Steven Ayre]

   1. recv 369 bytes from udp/[202.126.41.108]:5060 at 22:13:09.398177:
   2.
    ------------------------------------------------------------------------
   3.    SIP/2.0 604 Does not exist anywhere


The NO_ROUTE_DESTINATION clearing cause is due to the 604 sent to you from
your SIP provider. It is unrelated to your profile.

Only your SIP provider can tell you the reason for the 604 - raise it with
them. It could be a number of things - an invalid number, unreachable
number, internal/upstream error etc.

-Steve



On 12 May 2013 23:37, Steven Ayre <steveayre at gmail.com> wrote:

> Not sure why you replaced the gateway name with xxxx - it makes it
> impossible to see which gateway you used without looking through the log.
> It isn't hiding anything of value.
>
> As I said on my previous email, but in more detail:
>
> EXECUTE sofia/internal/407 at 10.15.1.41bridge(sofia/gateway/globe/0018444485452)
>
>
> You are dialing out via gateway 'globe'
>
> external::globe       gateway                sip:029080340 at siphosted.com     REGED
>
>
> The 'globe' gateway is configured on the 'external' profile.
>
> Any call sent OUT via the globe gateway will leave from the profile it is
> configured on.
>
> If your call comes IN via internal, it will go OUT via external since the
> gateway is on that profile. FS lets you dial between profiles (and even
> different protocols) so what you came in on does not affect what you leave
> on.
>
> This would be the expected and indeed required behaviour, since the
> gateway configured on external uses its IP and the siphosted.com provider
> may not be reachable from the IP used on other profiles.
>
> -Steve
>
>
>
>
> On 11 May 2013 06:43, baskar <yudha2008 at gmail.com> wrote:
>
>> Hi Steve,
>>
>> Thanks for the reply.
>>
>> When i try to use only the gateway it pass through external profile why it
>> is happning.
>>
>> SIP trunk has username and password with hard cable connected from service
>> provider.
>>
>> SIP trunk configured in freeswitch server and pass through service
>> provider
>> through SIP trunk.
>>
>> sofia/internal/407 at 10.15.1.41  bridge(sofia/gateway/xxxx/0018444485452)
>> 2013-05-08 06:13:09.376846 [NOTICE] switch_channel.c:669 New Channel
>> sofia/external/0018444485452 [4cdd9992-b763-11e2-bdef-2d40d364119f]
>>
>> Can any one guide me procedure to calls passing through the SIP server.
>>
>> Thanks in advance.
>>
>> Regards,
>> N.Baskar
>>
>>
>>
>> --
>> View this message in context:
>> http://freeswitch-users.2379917.n2.nabble.com/NO-ROUTE-DESTINATION-tp7590466p7590575.html
>> Sent from the freeswitch-users mailing list archive at Nabble.com.
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> 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/20130512/63af0390/attachment.html