[Freeswitch-users] Response status from client
Steven Ayre
steveayre at gmail.com
Thu Jun 9 14:52:09 MSD 2011
It's when it's executing the Local_Extension extension on
http://pastebin.freeswitch.org/16464 lines 741-772:
<extension name="Local_Extension">
<condition field="destination_number" expression="(^\d{2,7}$)">
<action application="set" data="dialed_extension=$1"/>
<action application="export" data="dialed_extension=$1"/>
<!-- bind_meta_app can have these args <key> [a|b|ab] [a|b|o|s]<app> -->
<action application="bind_meta_app" data="1 b s
execute_extension::dx XML features"/>
<action application="bind_meta_app" data="2 b s record_session::$${
recordings_dir}/archive/${strftime(%Y)}/${strftime(%b)}/${strftime(%d)}/${
uuid}.wav"/>
<action application="bind_meta_app" data="3 b s
execute_extension::cf XML features"/>
<action application="set" data="ringback=${us-ring}"/>
<action application="set" data="transfer_ringback=$${hold_music}"/>
<action application="set" data="call_timeout=30"/>
<!-- <action application="set" data="sip_exclude_contact=${
network_addr}"/> -->
<action application="set" data="hangup_after_bridge=true"/>
<!--<action application="set"
data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,USER_BUSY,NO_ANSWER,TIMEOUT,NO_ROUTE_DESTINATION"/>
-->
<action application="set" data="continue_on_fail=true"/>
<action application="hash" data="insert/${domain_name}-call_return/$
{dialed_extension}/${caller_id_number}"/>
<action application="hash" data="insert/${domain_name}
-last_dial_ext/${dialed_extension}/${uuid}"/>
<action application="set" data="called_party_callgroup=${user_data($
{dialed_extension}@${domain_name} var callgroup)}"/>
<!--<action application="export" data="nolocal:sip_secure_media=${
user_data(${dialed_extension}@${domain_name} var sip_secure_media)}"/>-->
<action application="hash" data="insert/${domain_name}-last_dial/${
called_party_callgroup}/${uuid}"/>
<action application="bridge" data="user/${dialed_extension}@${
domain_name}"/>
<action application="answer"/>
<action application="sleep" data="1000"/>
<!--<action application="bridge"
data="loopback/app=voicemail:default ${domain_name} ${dialed_extension}
"/>-->
<action application="voicemail" data="default ${domain_name} ${
dialed_extension}"/>
</condition>
</extension>
http://pastebin.freeswitch.org/16463 line 327 shows it's this extension
-Steve
On 9 June 2011 11:03, Alessandro <a.luppi at seletech.com> wrote:
> Hi,
>
> This is de default diaplan:
> http://pastebin.freeswitch.org/16464
>
> but I think that sofia read public dial plan, ths is the public dial plan:
>
> http://pastebin.freeswitch.org/16465
>
>
> I don't see the point where the FS read operation reported at lines
> 342-343.
>
> Thanks
>
> Regards
>
> Alessandro
>
>
>
> Il 09/06/2011 10:16, Steven Ayre ha scritto:
>
> It's a mistake in your dialplan. See lines 342-343:
>
> 1. Dialplan: sofia/internal/1009 at 192.168.2.101 Action bridge(user/${
> dialed_extension}@${domain_name})
> 2. Dialplan: sofia/internal/1009 at 192.168.2.101 Action answer()
> 3. Dialplan: sofia/internal/1009 at 192.168.2.101 Action sleep(1000)
> 4. Dialplan: sofia/internal/1009 at 192.168.2.101 Action voicemail(default
> ${domain_name} ${dialed_extension})
>
>
> You're answering the call after the bridge before it goes to voicemail.
> That answer is what is generating the 200.
>
> See lines 507-508 to see it happening:
>
> 1. 2011-06-09 09:15:15.627871 [INFO] mod_dptools.c:2393 Originate
> Failed. Cause: CALL_REJECTED
> 2. EXECUTE sofia/internal/1009 at 192.168.2.101 answer()
>
> -Steve
>
>
>
> On 9 June 2011 09:11, Steven Ayre <steveayre at gmail.com> wrote:
>
>> I don't see the siptrace in that log?
>>
>> -Steve
>>
>>
>> On 9 June 2011 08:35, Alessandro <a.luppi at seletech.com> wrote:
>>
>>> Hi,
>>>
>>> the url is: http://pastebin.freeswitch.org/16463
>>> i made a call from phone1 to phone2, the called party refused the call
>>> with code 603. FS received the status 603 form the called (softphone 2)
>>> party. Than FS sent to the calling party (softphone 1) the message 200 and
>>> bye.
>>>
>>> This is the resume of the log:
>>>
>>>
>>> 1000 at localnet_ip FS(ip:localnet_ip)
>>> 1001 at localnet_ip
>>>
>>> INVITE ---------->
>>> INVITE --------------->
>>> <-------------- trying <--------------------trying
>>> <------------------ 603
>>>
>>> <-------------- 200
>>>
>>> ACK------------------> <--------------------BYE
>>>
>>>
>>>
>>> *You said you had voicemail before... you can't send 603 back to the
>>> client and continue to voicemail because the 603 terminates the call.*
>>>
>>> When the called party terminates the call before answering, the calling
>>> party receive e registered message like "The phone called is not available,
>>> leave a message ...". Than i found the registered message in freeswitch.
>>> (I'm using fusion-pbx)
>>>
>>>
>>> Thanks
>>>
>>> Regards
>>>
>>> Alessandro
>>>
>>>
>>> Il 08/06/2011 21:55, Steven Ayre ha scritto:
>>>
>>> Question 1:
>>>> i'm developing a custom client sip with pjsip. This client when receive
>>>> a call that can't be accepted respond with status 603. I think that
>>>> freeswitch filter this status.
>>>>
>>>
>>> 603 gets treated fine for me. I think we need to see more information -
>>> can you put a debug level log of the call with siptrace enabled (sofia
>>> global siptrace on) on pastebin (http://pastebin.freeswitch.org/) and
>>> then post the url here?
>>>
>>> Chances are you're doing something in the dialplan that's answering the
>>> call, either before or after the failed bleg.
>>>
>>> You said you had voicemail before... you can't send 603 back to the
>>> client and continue to voicemail because the 603 terminates the call.
>>>
>>>
>>>> Question:2
>>>>
>>> It's possible a custom Header pass trough in status response like trying
>>>> or session in progress? I'm able to use custom header only on invite adding
>>>> to invite a header with name like X-myheader. Any suggestion?
>>>>
>>>
>>> Yes, you can for 180/183, with the sip_ph_X- prefix. That puts the header
>>> on any provisional response.
>>> http://wiki.freeswitch.org/wiki/Sofia#Adding_Response_Headers
>>>
>>> For example:
>>> <action application="set" data="sip_p_X-myheader=value"/>
>>>
>>> AFAIK you won't be able to do the same for a 100 Trying since Sofia
>>> doesn't let FS do any handling at the required point. But have a try anyway
>>> just to be sure.
>>>
>>> -Steve
>>>
>>>
>>> On 8 June 2011 20:14, Alessandro <a.luppi at seletech.com> wrote:
>>>
>>>> Hi,
>>>>
>>>> I have two questions about FS:
>>>>
>>>> Question 1:
>>>> i'm developing a custom client sip with pjsip. This client when receive
>>>> a call that can't be accepted respond with status 603. I think that
>>>> freeswitch filter this status.
>>>> This is an example of desired behaviour:
>>>> 1000 at localnet_ip FS(ip:localnet_ip)
>>>> 1001 at localnet_ip
>>>>
>>>> INVITE ---------->
>>>> INVITE --------------->
>>>> <-------------- trying <--------------------trying
>>>> <-------------- 603 <------------------ 603
>>>> ACK ------------> ACK------------------>
>>>>
>>>>
>>>>
>>>> The current behaviour of FS is:
>>>>
>>>> 1000 at localnet_ip FS(ip:localnet_ip)
>>>> 1001 at localnet_ip
>>>>
>>>> INVITE ---------->
>>>> INVITE --------------->
>>>> <-------------- trying <--------------------trying
>>>> <------------------ 603
>>>>
>>>> <-------------- 200
>>>>
>>>> ACK------------------> <--------------------BYE I'd like to avoid the
>>>> current behaviour. It's possible a kind of message status path trough? If
>>>> the called party terminate the call before answering, FS send always to the
>>>> calling partner 200 and BYE. First thought was related to the voice-mail.
>>>> Now voice-mail is disabled but the behaviour is the same. Question:2 It's
>>>> possible a custom Header pass trough in status response like trying or
>>>> session in progress? I'm able to use custom header only on invite adding to
>>>> invite a header with name like X-myheader. Any suggestion? Thanks Good
>>>> Evening
>>>>
>>>> Alessandro
>>>>
>>>> --
>>>> Ing. Alessandro Luppi
>>>> Software development
>>>> Seletech srl
>>>> Via Collodi 8, 20052 Monza (MI) - Italy
>>>> Tel: +39.039.5962000 - Fax: +39.039.9716905
>>>> email: a.luppi at seletech.com - Web: www.seletech.com or www.seletech.eu
>>>>
>>>>
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>>>>
>>>
>>> _______________________________________________
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>>>
>>>
>>>
>>> --
>>> Ing. Alessandro Luppi
>>> Software development
>>> Seletech srl
>>> Via Collodi 8, 20052 Monza (MI) - Italy
>>> Tel: +39.039.5962000 - Fax: +39.039.9716905
>>> email: a.luppi at seletech.com - Web: www.seletech.com or www.seletech.eu
>>>
>>>
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>>>
>>
>
> _______________________________________________
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>
>
>
> --
> Ing. Alessandro Luppi
> Software development
> Seletech srl
> Via Collodi 8, 20052 Monza (MI) - Italy
> Tel: +39.039.5962000 - Fax: +39.039.9716905
> email: a.luppi at seletech.com - Web: www.seletech.com or www.seletech.eu
>
>
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