[Freeswitch-users] Answer issue on inbound call
Max Alex
max.asterisk at gmail.com
Fri Jul 1 08:12:15 MSD 2011
Hi,
Can any one provide their suggestions and help for this issue,
I really need to resolve and get this working.
Thanks,
Max Alex
Voip Developer
On Tue, Jun 28, 2011 at 6:01 PM, Max Alex <max.asterisk at gmail.com> wrote:
> Hi,
> Thanks for your reply.
> I have enabled logger as per your help.
> I have given completed log on following link of pastebin.
> http://pastebin.freeswitch.org/16616
>
> You can see this line as it is showing answered and this call is answered
> on my cell phone but on softphone it is ringing and i have rejected from
> there.
> 2011-06-28 17:47:52.909657 [NOTICE] mod_freetdm.c:1953 Channel [FreeTDM/2:
> 1/01234567890] has been answered
>
> It is pre answering the cell phone when it is ringing on phone
> Waiting for your help.
>
> Thanks,
> Max Alex
> Voip Developer
>
>
>
> On Mon, Jun 27, 2011 at 9:15 PM, Michael Collins <msc at freeswitch.org>wrote:
>
>> Get a complete, unedited, unabridged console debug log w/ siptrace and put
>> it on pastebin w/ "FreeSWITCH Log" for the syntax highlighting. Use "sofia
>> global siptrace on" to make sure you can all SIP traffic.
>>
>> -MC
>>
>>
>> On Mon, Jun 27, 2011 at 6:05 AM, Max Alex <max.asterisk at gmail.com> wrote:
>>
>>> Hi,
>>> Thanks for reply,
>>> I have tried the same way and reloaded freeswitch, but still it is
>>> answered on first ring of the call.
>>> When it is ringing the call on 1001 and the same time it is answered on
>>> cell phone, so something is done when it is ringing on 1001.
>>>
>>> Here is the dialplan for the same
>>>
>>> <extension name="Local_Extension">
>>> <condition field="destination_number" expression="^(10[01][0-9])$">
>>> <action application="set" data="dialed_extension=$1"/>
>>> <action application="export" data="dialed_extension=$1"/>
>>> <action application="bind_meta_app" data="2 b s
>>> record_session::$${recordings_dir}/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>
>>> <action application="bind_meta_app" data="3 b s
>>> execute_extension::cf XML features"/>-->
>>> <action application="set" data="ringback=${us-ring}"/>
>>> <action application="set"
>>> data="transfer_ringback=$${hold_music}"/>
>>> <action application="set" data="call_timeout=30"/>
>>> <action application="set" data="hangup_after_bridge=true"/>
>>> <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}@${domain_name}
>>> var callgroup)}"/>
>>> <action application="hash"
>>> data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/>
>>>
>>> <action application="bridge"
>>> data="{ignore_early_media=true}user/${dialed_extension}@
>>> ${domain_name}"/>
>>> <action application="sleep" data="1000"/>
>>> <action application="voicemail" data="default ${domain_name}
>>> ${dialed_extension}"/>
>>> </condition>
>>> </extension>
>>>
>>>
>>> Please help me for this issue.
>>>
>>>
>>> Thanks,
>>> Max Alex
>>> Voip Developer
>>>
>>>
>>>
>>> On Fri, Jun 24, 2011 at 11:38 AM, Michael Collins <msc at freeswitch.org>wrote:
>>>
>>>> Are you using the default dialplan? I think you might just need to
>>>> ignore early media on your bridge app. If you are using the default.xml file
>>>> then locate "Local_Extension" and the bridge line:
>>>>
>>>> <action application="bridge" data="user/${dialed_extension}@
>>>> ${domain_name}"/>
>>>>
>>>> Change it to this, then try again:
>>>>
>>>> <action application="bridge"
>>>> data="{ignore_early_media=true}user/${dialed_extension}@
>>>> ${domain_name}"/>
>>>>
>>>> If I understand correctly, the "symptom" you are experiencing is the
>>>> normal operation of the bridge app (and it's cousin, the originate API
>>>> command). When the b-leg sends back media, including ringing, the bridge (or
>>>> the originate) will be considered "successful," and in the case of bridge,
>>>> the b-leg's audio (the early media) will be connected to the a-leg. If you
>>>> set ignore_early_media=true then you are explicitly telling the bridge app
>>>> that you only want to connect the b-leg to the a-leg if the b-leg actually
>>>> answers.
>>>>
>>>> I hope that made sense...
>>>>
>>>> -MC
>>>>
>>>>
>>>>
>>>> On Thu, Jun 23, 2011 at 9:32 PM, Max Alex <max.asterisk at gmail.com>wrote:
>>>>
>>>>> Hi,
>>>>> Thanks for reply.
>>>>> Current scenario is below.
>>>>>
>>>>> PSTN call -> sangoma device -> freeswitch incoming context -> routed to
>>>>> 1001 -> ringing (Answered on cellphone)
>>>>> Here when it is routed to 1001 the call it is started ringing, but on
>>>>> phone that call is answered and hearding sound of ringing.
>>>>>
>>>>> Required flow:
>>>>> PSTN call -> sangoma device -> freeswitch incoming context -> routed to
>>>>> 1001 -> ringing (Ringing on cellphone)
>>>>>
>>>>> I hope it is understable, the call should not be answered until 1001
>>>>> answer it, right not when it is started ring it is answered on cell phone.
>>>>> that should not be happen as it is not answered yet.
>>>>>
>>>>> Waiting for your reply.
>>>>>
>>>>>
>>>>> Thanks,
>>>>> Max Alex
>>>>> Voip Developer
>>>>>
>>>>>
>>>>>
>>>>> On Fri, Jun 24, 2011 at 5:48 AM, Michael Collins <msc at freeswitch.org>wrote:
>>>>>
>>>>>> I'm not sure I understand the problem. What is happening vs. what you
>>>>>> believe should be happening?
>>>>>> -MC
>>>>>>
>>>>>>
>>>>>> On Thu, Jun 23, 2011 at 3:31 AM, Max Alex <max.asterisk at gmail.com>wrote:
>>>>>>
>>>>>>> Hi,
>>>>>>> Thanks for your reply.
>>>>>>> Here is my configuration and log
>>>>>>> http://pastebin.freeswitch.org/16571
>>>>>>>
>>>>>>> I am using A200 analog sangoma device with freeswitch, it is working
>>>>>>> fine but when it is routing call to 1001 then it is answered.
>>>>>>> Please provider your suggestions.
>>>>>>>
>>>>>>> Thanks,
>>>>>>> Max Alex
>>>>>>> Voip Developer
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> On Wed, Jun 22, 2011 at 8:51 PM, Michael Collins <msc at freeswitch.org
>>>>>>> > wrote:
>>>>>>>
>>>>>>>> I thought the A200 was an analog card? Maybe I have my numbers mixed
>>>>>>>> up...
>>>>>>>>
>>>>>>>> Go ahead and collect a debug log of this call. It might help to have
>>>>>>>> your configs posted as well. Use pastebin.freeswitch.org. See this
>>>>>>>> wiki article for tips on how to collect information:
>>>>>>>> http://wiki.freeswitch.org/wiki/Reporting_Bugs
>>>>>>>>
>>>>>>>> -MC
>>>>>>>>
>>>>>>>> On Wed, Jun 22, 2011 at 3:23 AM, Max Alex <max.asterisk at gmail.com>wrote:
>>>>>>>>
>>>>>>>>> Hi,
>>>>>>>>> I have installed freeswitch latest version with sangoma card A200
>>>>>>>>> as well,
>>>>>>>>> I have installed and configured freetdm module with wanpipe drivers
>>>>>>>>> for freeswitch,
>>>>>>>>> We are properly receiving the inbound calls in public context and
>>>>>>>>> then we are routing that call to 1001 extension,
>>>>>>>>> it is properly routing to 1001 as well, but we have one issue while
>>>>>>>>> routing on 1001.
>>>>>>>>>
>>>>>>>>> Here is the issue description.
>>>>>>>>> I am calling from my cell phone to that DID number of pri line, and
>>>>>>>>> then it will start ringing on 1001 extension,
>>>>>>>>> When 1001 extension start ringing the call is answered on my cell
>>>>>>>>> phone,
>>>>>>>>> it is something like freeswitch preanswer or autoanswer the call,
>>>>>>>>> how can i stop this answer call when it is ringing on 1001 extension,
>>>>>>>>> Waiting for good reply.
>>>>>>>>>
>>>>>>>>> Thanks,
>>>>>>>>> Max Alex
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> _______________________________________________
>>>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>>>>
>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
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>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>
>>>>>>>>>
>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>>>
>>>>>>>> FreeSWITCH-users mailing list
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>>>>>>>>
>>>>>>>>
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>>
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>>>>>>>
>>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>
>>>>>> FreeSWITCH-users mailing list
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>>>>>>
>>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>
>>>>> FreeSWITCH-users mailing list
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>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>> http://www.cluecon.com 877-7-4ACLUE
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>>>>
>>>
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com 877-7-4ACLUE
>>>
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>>>
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
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>>
>>
>
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