[Freeswitch-users] Answer issue on inbound call

Michael Collins msc at freeswitch.org
Fri Jul 1 20:21:06 MSD 2011 

What's with the errors on the voicemail sound files?
-MC

On Thu, Jun 30, 2011 at 9:12 PM, Max Alex <max.asterisk at gmail.com> wrote:

> 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
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>>>>>>>>>>
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>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>
>>>>>>>>> _______________________________________________
>>>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
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>>>>>>>>>
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>>>>>>>>>
>>>>>>>>>
>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
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>>>>>>>>
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>>>>>>>>
>>>>>>>
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>>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>
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>>>>>>
>>>>>
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>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>> http://www.cluecon.com 877-7-4ACLUE
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>>>>>
>>>>
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>>>
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>>
>
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