[Freeswitch-users] conference from a sip provider

[Antony King]

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 at 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 at 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 at 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 at 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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