Hi All, I have been testing freeswitch for more than a year. But now i am facing problem in new configuartion of SIP trunk. Here i have pasted by log http://pastebin.freeswitch.org/20882. When i try to call through gateway I execute through external profile. EXECUTE sofia/internal/407@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] My dialplan default.xml <extension name="xxxx"> <condition field="destination_number" expression="2(\d+)$"> <action application="set" data="continue_on_fail=CALL_REJECTED,GATEWAY_DOWN,RECOVERY_ON_TIMER_EXPIRE,NORMAL_TEMPORARY_FAILURE,DESTINATION_OUT_OF_ORDER,NORMAL_UNSPECIFIED"/> <action application="set" data="hangup_after_bridge=true"/> <action application="bridge" data="sofia/gateway/xxxx/001$1"/> </condition> </extension> Could some one help where i am wrong. Thanks in advance. Regards, N.Baskar -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/NO-ROUTE-DESTINATION-tp7590466... Sent from the freeswitch-users mailing list archive at Nabble.com.
A call to a gateway will leave from the Sofia profile that the gateway is configured on. -Steve On 8 May 2013 11:28, baskar <yudha2008@gmail.com> wrote:
Hi All,
I have been testing freeswitch for more than a year. But now i am facing problem in new configuartion of SIP trunk.
Here i have pasted by log http://pastebin.freeswitch.org/20882.
When i try to call through gateway I execute through external profile.
EXECUTE sofia/internal/407@10.15.1.41 bridge(sofia/gateway/xxxx/0018444485452) 2013-05-08 06:13:09.376846 [NOTICE] switch_channel.c:669 New Channel sofia/external/0018444485452 [4cdd9992-b763-11e2-bdef-2d40d364119f]
My dialplan default.xml
<extension name="xxxx"> <condition field="destination_number" expression="2(\d+)$"> <action application="set" data="continue_on_fail=CALL_REJECTED,GATEWAY_DOWN,RECOVERY_ON_TIMER_EXPIRE,NORMAL_TEMPORARY_FAILURE,DESTINATION_OUT_OF_ORDER,NORMAL_UNSPECIFIED"/> <action application="set" data="hangup_after_bridge=true"/> <action application="bridge" data="sofia/gateway/xxxx/001$1"/> </condition> </extension>
Could some one help where i am wrong.
Thanks in advance.
Regards, N.Baskar
-- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/NO-ROUTE-DESTINATION-tp7590466... Sent from the freeswitch-users mailing list archive at Nabble.com.
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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@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-tp7590466... Sent from the freeswitch-users mailing list archive at Nabble.com.
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@10.15.1.41bridge(sofia/gateway/globe/0018444485452) You are dialing out via gateway 'globe' external::globe gateway sip:029080340@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@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@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-tp7590466... Sent from the freeswitch-users mailing list archive at Nabble.com.
_________________________________________________________________________ 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
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@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@10.15.1.41bridge(sofia/gateway/globe/0018444485452)
You are dialing out via gateway 'globe'
external::globe gateway sip:029080340@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@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@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-tp7590466... Sent from the freeswitch-users mailing list archive at Nabble.com.
_________________________________________________________________________ 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
participants (2)
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baskar
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Steven Ayre