[Freeswitch-users] outbound sip call: NO_USER_RESPONSE

[Neil Patel]

As I suspected, the problem was a silly one. Extension 5001 is reserved by
dynamic_conference in the default dialplan. Once I changed to an unused
extension it works fine.

Nandy, once I bridge a call, how do I stream audio to the external number?
When I try session:streamfile (after checking session:ready() == true), the
file plays to the initiating FS extension, but not out to the external
number. Any suggestions?

Thanks,
Neil

On Fri, Apr 23, 2010 at 4:27 PM, Nandy Dagondon <nandy1925 at gmail.com> wrote:

> neil,
>
> in makecall.lua, you're creating a new session. don't. when makecall.lua is
> called, it's already in a session.  try this one:
>     session:execute("bridge","sessiondata")
>
> for flexibility,  it's better to pass the URI to the lua script at the
> dialplan.
>
> -nandy
>
>
> On Fri, Apr 23, 2010 at 6:16 AM, Neil Patel <neilp at cs.stanford.edu> wrote:
>
>> Thanks for clarifying.
>>
>> Here's the output (with siptrace on) when trying to dial out over the
>> gateway:
>>
>> 2010-04-22 15:13:10.104866 [NOTICE] switch_channel.c:669 New Channel
>> sofia/internal/1001 at server.com [a6aac86b-4a40-41f3-8968-f9b37501483a]
>> 2010-04-22 15:13:10.112850 [INFO] mod_dialplan_xml.c:418 Processing
>> neilp->5001 in context default
>> 2010-04-22 15:13:10.120874 [INFO] switch_core_session.c:1750 Sending early
>> media
>> 2010-04-22 15:13:10.122856 [NOTICE] mod_sofia.c:1907 Pre-Answer
>> sofia/internal/1001 at server.com!
>> 2010-04-22 15:13:10.123810 [NOTICE] switch_channel.c:669 New Channel
>> sofia/internal/1234 at conference.freeswitch.org[af13cf37-eb6a-4e9f-b1cc-366b6a24aec5]
>>  2010-04-22 15:13:10.386879 [INFO] switch_rtp.c:2049 Auto Changing port
>> from IP1 to IP2
>> 2010-04-22 15:13:10.697870 [NOTICE] sofia.c:4789 Hangup sofia/internal/
>> 1234 at conference.freeswitch.org [CS_CONSUME_MEDIA] [NO_USER_RESPONSE]
>> 2010-04-22 15:13:10.706880 [ERR] mod_conference.c:4563 Cannot create
>> outgoing channel, cause: NO_USER_RESPONSE
>> 2010-04-22 15:13:10.706880 [NOTICE] mod_conference.c:4566 Hangup
>> sofia/internal/1001 at server.com [CS_EXECUTE] [NO_USER_RESPONSE]
>> 2010-04-22 15:13:10.706880 [NOTICE] switch_core_session.c:1179 Session 2
>> (sofia/internal/1234 at conference.freeswitch.org) Ended
>> 2010-04-22 15:13:10.706880 [NOTICE] switch_core_session.c:1181 Close
>> Channel sofia/internal/1234 at conference.freeswitch.org [CS_DESTROY]
>> 2010-04-22 15:13:10.714854 [NOTICE] switch_core_session.c:1179 Session 1
>> (sofia/internal/1001 at server.com) Ended
>> 2010-04-22 15:13:10.714854 [NOTICE] switch_core_session.c:1181 Close
>> Channel sofia/internal/1001 at server.com [CS_DESTROY]
>>
>>
>>  In case it is relevant, the call is triggered in a lua script which is
>> executed by dialing a extension 5001. The dialplan in
>> freeswitch/conf/dialplan/default/5001_gizmo.xml:
>>  <extension name="gizmo">
>>       <condition field="destination_number" expression="^5001$">
>>       <action application="lua" data="makecall.lua"/>
>>     </condition>
>>   </extension>
>>
>> makecall.lua:
>> function make_call()
>>   sessiondata = "sofia/gateway/gizmo/<num>"
>>    new_session = freeswitch.Session(sessiondata)
>> end
>> make_call()
>>
>>
>>
>> On Thu, Apr 22, 2010 at 2:58 PM, William Suffill <
>> william.suffill at gmail.com> wrote:
>>
>>> gizmo is your gateway.
>>>
>>>
>>> siptrace is by profile.
>>>
>>>
>>> sofia status in the cli will list your profiles/gateways (default cfgs
>>> use internal/external)
>>>
>>> so in that case sofia profile external siptrace on
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