[OpenSIPS-Users] call a standard phone and mobile at the same time

[Muhammad Shahzad]

RTFM dude!

http://www.opensips.org/Resources/DocsCoreRoutes18

Thank you.


On Thu, Oct 11, 2012 at 6:23 PM, Engineer voip <forvoip4 at gmail.com> wrote:

> Hi,
> Thank you all
>  But i thought to do this by hanging up the call of mobile after some time if
> the mobile phone does not respond
> Is opensips can do that?
> thank you for your help
>
> 2012/10/9 Schneur Rosenberg <rosenberg11219 at gmail.com>
>
>> That's true but from my experience it worked for me with a few carriers,
>> of course every provider might treat it differently, even those that played
>> a message did not actually pick up the call it was done with early media.
>>  On Oct 9, 2012 2:55 PM, "Adrian Georgescu" <ag at ag-projects.com> wrote:
>>
>>> You cannot control the behavior of the GSM operators. They may decide to
>>> play a prompt and practically answer the call without returning any code
>>> when the target device is out of coverage, voicemail box may kick in or
>>> not, roaming or other scenarios you are not even aware about.  If instead
>>> of a response code, you end-up being connected or in early media, OpenSIPS
>>> cannot retarget the call in all situations. Thinks work random and fail for
>>> unknown reasons due to un-deterministic  behavior of the GSM operators
>>> involved.
>>>
>>> Adrian
>>>
>>> On Oct 9, 2012, at 2:40 PM, Schneur Rosenberg wrote:
>>>
>>> If you disable voicemail on the mobile phone through the carrier then
>>> you should get a busy signal from the carrier, if you also disable call
>>> waiting then you should also get busy a busy signal from the carrier, its
>>> not really required if you don't mind being disturbed by the beep , this
>>> might not always be a solution because you loose voicemail and call waiting
>>> on your phone.
>>>
>>> Another solution might be to send the call through a Asterisk server and
>>> use the amd (answering machine detection) function.
>>> On Oct 8, 2012 11:17 PM, "Adrian Georgescu" <ag at ag-projects.com> wrote:
>>>
>>>> There is no standard for communicating signaling information from a
>>>> non-IP network like GSM to a SIP based network. What you try to achieve is
>>>> close to impossible unless you own both networks and are able to set up
>>>> some proprietary signaling mechanism between them.
>>>>
>>>> Adrian
>>>>
>>>> On Oct 8, 2012, at 10:56 PM, Engineer voip wrote:
>>>>
>>>> Hi,
>>>> I want to associate a mobile phone number to each desktop phone and
>>>> want to ringing at the same time the desktop phone and mobile phone:
>>>> ** if the mobile phone is off service or already in communication (call
>>>> falls on the responder) the opensips ignore this mobile and keep ringing
>>>> the desktop phone.
>>>>
>>>> i hope this is clear
>>>>
>>>> 2012/10/8 Duane Larson <duane.larson at gmail.com>
>>>>
>>>>> Can you explain in more detail what you are trying to do?
>>>>>
>>>>>
>>>>> On Mon, Oct 8, 2012 at 8:32 AM, Engineer voip <forvoip4 at gmail.com>wrote:
>>>>>
>>>>>> Hello,
>>>>>> I do that and its good but if user A hangs up the call, the user B is
>>>>>> hanging too.
>>>>>> I want when user A hangs up or responder,  the user B can take the
>>>>>> call!
>>>>>> have you an idea to resolve that?
>>>>>>
>>>>>> 2012/10/8 Vlad Paiu <vladpaiu at opensips.org>
>>>>>>
>>>>>>> **
>>>>>>> Hello,
>>>>>>>
>>>>>>> This is the usual SIP parallel forking, so yes this can be done with
>>>>>>> OpenSIPS 1.8. See the append_branch() function at [1] .
>>>>>>>
>>>>>>> [1] http://www.opensips.org/Resources/DocsCoreFcn18#toc106
>>>>>>>
>>>>>>> Regards,
>>>>>>>
>>>>>>> Vlad Paiu
>>>>>>> OpenSIPS Developerhttp://www.opensips-solutions.com
>>>>>>>
>>>>>>>
>>>>>>> On 10/08/2012 12:00 AM, Engineer voip wrote:
>>>>>>>
>>>>>>> Hi All,
>>>>>>> It is possible to call a standard (fix) phone and mobile phone atthe same time
>>>>>>> with OpenSIPS 1.8 ??
>>>>>>>
>>>>>>> --
>>>>>>>
>>>>>>> Best Regards.
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>>>>>
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> Users mailing list
>>>>>>> Users at lists.opensips.org
>>>>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>>>>>
>>>>>>>
>>>>>>
>>>>>>
>>>>>> --
>>>>>>
>>>>>> Best Regards.
>>>>>>
>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> Users mailing list
>>>>>> Users at lists.opensips.org
>>>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>>>>
>>>>>>
>>>>>
>>>>>
>>>>> --
>>>>> --
>>>>> *--*--*--*--*--*
>>>>> Duane
>>>>> *--*--*--*--*--*
>>>>> --
>>>>>
>>>>> _______________________________________________
>>>>> Users mailing list
>>>>> Users at lists.opensips.org
>>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>>>
>>>>>
>>>>
>>>>
>>>> --
>>>>
>>>> Best Regards.
>>>>
>>>>
>>>> _______________________________________________
>>>> Users mailing list
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>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> Users mailing list
>>>> Users at lists.opensips.org
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>>>>
>>>> _______________________________________________
>>> Users mailing list
>>> Users at lists.opensips.org
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>>>
>>>
>>>
>>> _______________________________________________
>>> Users mailing list
>>> Users at lists.opensips.org
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>>>
>>>
>> _______________________________________________
>> Users mailing list
>> Users at lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
>
>
> --
>
> Best Regards.
>
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>


-- 
Muhammad Shahzad
-----------------------------------
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +92 334 422 40 88
MSN: shari_786pk at hotmail.com
Email: shaheryarkh at googlemail.com
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