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[Freeswitch-users] conference from a sip provider


 
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antony.king at solutio...
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PostPosted: Wed Apr 15, 2009 10:31 am    Post subject: [Freeswitch-users] conference from a sip provider Reply with quote

I'm just getting started with freeswitch; I'd like to create a public phone number that a small number of people can dial into to join a conference.

I've got calls and the conference rooms working internally, and I've got a link to my sipgate account which directs to extension 1000 . The configuration is virtually unchanged from the default. I created sip_profiles/external/sipgate.xml using the default template in that folder and put my details in.

However, if I change sip_profiles/external/sipgate.xml to point to 3001, ie

<param name="extension" value="3001"/>

then when the external call comes in, freeswitch does this 3 times then stops:


2009-04-15 16:05:30 [NOTICE] switch_channel.c:567 switch_channel_set_name() New Channel sofia/external/0XXXXXXXXX@sipgate.co.uk [e56ac315-62e8-40b8-aa5a-bcc5dbd841ee]

2009-04-15 16:05:30 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing 0XXXXXXXXX->3001 in context public

2009-04-15 16:05:30 [NOTICE] switch_core_state_machine.c:168 switch_core_standard_on_execute() Hangup sofia/external/0XXXXXXX@sipgate.co.uk [CS_EXECUTE] [NORMAL_CLEARING]

2009-04-15 16:05:30 [NOTICE] switch_core_session.c:970 switch_core_session_thread() Session 1 (sofia/external/0XXXXXXX@sipgate.co.uk) Ended

2009-04-15 16:05:30 [NOTICE] switch_core_session.c:972 switch_core_session_thread() Close Channel sofia/external/0XXXXXXX@sipgate.co.uk [CS_HANGUP]

Presumably there's some difference between calls coming in via a gateway and localy generated calls; could someone give me some pointers as to how to get it to accept the call ?
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brian at freeswitch.org
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PostPosted: Wed Apr 15, 2009 10:40 am    Post subject: [Freeswitch-users] conference from a sip provider Reply with quote

Press F8 and try again... With debug cranked up you'll see more details.
On Apr 15, 2009, at 10:10 AM, Antony King wrote:
Quote:
sumably there's some difference between calls coming in via a gateway and localy generated calls; could someone give me some pointers as to how to get it to accept the call ?


Brian West
brian@freeswitch.org (brian@freeswitch.org)



-- Meet us at ClueCon! http://www.cluecon.com
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antony.king at solutio...
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PostPosted: Thu Apr 16, 2009 4:25 am    Post subject: [Freeswitch-users] conference from a sip provider Reply with quote

Aha - hadn't seen that one.

This was in the log after pressing f8:

--
2009-04-16 09:50:58 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing 0XXXXXXXXXXX->3001 in context public
2009-04-16 09:50:58 [DEBUG] mod_dialplan_xml.c:117 parse_exten() Regex: [public_extensions] destination_number(3001) =~ /^(10[01][0-9])$/
2009-04-16 09:50:58 [DEBUG] mod_dialplan_xml.c:119 parse_exten() Regex mismatch
--

so I've put this in dialplan/public.xml:

<extension name="public_conferences">
<condition field="destination_number" expression="^(3[0-9][0-9][0-9])$">
<action application="transfer" data="$1 XML default"/>
</condition>
</extension>

Now it accepts the call:

--
2009-04-16 09:54:28 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing 0XXXXXXXXXXX->3001 in context public
2009-04-16 09:54:28 [DEBUG] mod_dialplan_xml.c:117 parse_exten() Regex: [public_conferences] destination_number(3001) =~ /^(3[0-9][0-9][0-9])$/
2009-04-16 09:54:28 [DEBUG] switch_core_state_machine.c:100 switch_core_standard_on_routing() (sofia/external/0XXXXXXXXXXX@sipgate.co.uk) State Change CS_ROUTING -> CS_EXECUTE
--

Don't get any audio at the moment, but I think that's a separate problem.

Thanks for the tip,

Antony.


On Wednesday 15 April 2009 16:24:45 Brian West wrote:
Quote:
Press F8 and try again... With debug cranked up you'll see more details.

On Apr 15, 2009, at 10:10 AM, Antony King wrote:
Quote:
sumably there's some difference between calls coming in via a
gateway and localy generated calls; could someone give me some
pointers as to how to get it to accept the call ?

Brian West
brian@freeswitch.org

-- Meet us at ClueCon! http://www.cluecon.com

--

Antony King - 01908 268 901
Systems Consultant
SolutionTrax Technologies - http://www.solutiontrax.com
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jason at jasonjgw.net
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PostPosted: Thu Apr 16, 2009 4:57 am    Post subject: [Freeswitch-users] conference from a sip provider Reply with quote

Antony King <antony.king@solutiontrax.com> wrote:
Quote:
so I've put this in dialplan/public.xml:

<extension name="public_conferences">
<condition field="destination_number" expression="^(3[0-9][0-9][0-9])$">
<action application="transfer" data="$1 XML default"/>
</condition>
</extension>

Wouldn't it be better to put it in dialplan/public/3xxx-conference.xml (or a
similar file name of your choice)? That way, you could leave public.xml
unmodified, and more easily manage the configuration.


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