[asterisk-users] Mobile Phones and Asterisk

jon pounder jonp at inline.net
Thu Oct 28 22:40:35 CDT 2010 

On 10/28/2010 11:18 PM, GBR Icasiano, Ryan A. wrote:

Here is what I do today and it works fine:

- asterisk/trixbox
- Dext/android phone
- Bell Canada cell provider
- call comes in, to an extension with voicemail
- rings a bunch of sip devices (real phones, and the android via 
linphone if it happens to be near wifi and registered (set to only use 
wifi not 3g to register)
- if not answered call is forwarded back out a pots line and dials the 
cell number (cell is not subscribed to provider voicemail)
- still no answer that pots line is hung up and call drops back into the 
original extension's vm. (I have not run into a problem with answer 
detection, only that people don't stay on the line long enough for me to 
answer on the second set of ringing, but if they are that impatient the 
call was probably not important anyway)

outgoing calls if registered I have a choice once I dial of linphone or 
dialer to make the call.

checking vm is just *98<ext> from linphone as the dialing app, or dial 
in and navigate to vm.

linphone is a little less polished gui but seems to work the best for me 
to reliably register when it should.
(tried about 5 different sip clients)




> Hi,
>
> Thanks for your very informative response. This is really helpful. I wouldn't be pushing it though since it isn't possible as of now.
>
> Kudos!
>
> RYAN ICASIANO
> ________________________________________
> From: asterisk-users-bounces at lists.digium.com [asterisk-users-bounces at lists.digium.com] On Behalf Of Sebastian [shop at open-t.co.uk]
> Sent: Friday, October 29, 2010 5:50 AM
> To: asterisk-users at lists.digium.com
> Subject: Re: [asterisk-users] Mobile Phones and Asterisk
>
> Hi,
>
> On 10/28/2010 11:20 AM, GBR Icasiano, Ryan A. wrote:
>    
>> Hi,
>>
>> I can actually place a successful call using that configuration. The telco i'm currently working requires the prefix.
>>
>> What I'm trying to do is to capture the status of the mobile phone, if it is currently engaged in a call or not.
>>      
> Maybe others who know better will jump in - but I seriously doubt you
> will be able to do this. From my limited knowledge, I believe mobile
> phone networks use different signalling then regular terrestrial based
> providers. I don't really think that the engaged tone sent back by the
> mobile operator will be decoded correctly by Asterisk.
>
> Not to mention that, I don't what happens where you are - but in UK for
> example - you don't even get an engaged tone from a mobile phone. You
> just get either sent to the user's voice mail, or you are played a
> message from the mobile phone operator which essentially tells you that
> the user is engaged or unavailable. Operators in many other European
> countries do the same. So from the point of what you are trying to
> achieve - this is useless in Asterisk.
>
> I would have liked to do the same thing - as I have line divert in
> Asterisk to my mobile phone - and I would have liked for Asterisk to
> just skip along to my Asterisk voice mail when my mobile is either out
> of coverage, or when I'm in a conversation on it. But no such luck. I
> believe the mobile operators wouldn't like the idea anyway - as they get
> to charge you extra for playing all those messages or sending you to
> their voicemail.
>
> I believe in parts of the North American continent things are similar,
> but even worse. As the caller gets charged as soon as the mobile phone
> starts ringing - apparently simply the act of accessing the mobile
> operator's network is chargeable - never mind if you get to speak to
> anybody or not.
>
> Then again, maybe things are different where you are - and maybe there
> is a way to get Asterisk to recognise the busy tone from your mobile
> operator. Maybe somebody here will jump in with a suggestion. It seems
> that it has to do with "busy signalling" in Asterisk. A softphone I
> believe will accomplish this out of band - with some commands over SIP.
> While PSTN (normal phone lines) and mobiles I believe tend to signal
> this with inband tones (part of the sound coming down the line).
>
> You might also want to check your regional settings in Asterisk.
>
>
> Sebastian
>
> I achieved this successfully by emulating it via a softphone, when I
> call a softphone and it is currently engaged in a call, asterisk returns
> BUSY in DIALSTATUS and will automatically fallback to the next step in
> the dialplan.
>    
>> But this is not the case when applying it to the mobile phone. When the target phone is currently engaged in a call, and I called the mobile phone, I can hear a "busy tone"(which is alright, since the target phone is actually busy), but it will wait until it timed out as defined in the DIAL cmd, and the "var DIALSTATUS" returns NOANSWER, instead of BUSY, as if the mobile phone is available and it was not answered at all.
>>
>> It may also have to do on how the tones are being handled, or it can also be that the mobile phone and the media gateway are the one talking to each other, and asterisk cannot get the status of the phone itself.
>>
>> regards,
>>
>> RYAN ICASIANO
>> ________________________________________
>> From: asterisk-users-bounces at lists.digium.com [asterisk-users-bounces at lists.digium.com] On Behalf Of Sebastian [shop at open-t.co.uk]
>> Sent: Thursday, October 28, 2010 5:27 PM
>> To: asterisk-users at lists.digium.com
>> Subject: Re: [asterisk-users] Mobile Phones and Asterisk
>>
>> Hi,
>>
>> On 10/28/2010 01:06 AM, GBR Icasiano, Ryan A. wrote:
>>      
>>> Hi,
>>>
>>> Thanks for your reply. I'm calling a normal phone using the DIAL cmd. Here is my sample dial command:
>>>
>>> exten =>s,4,Dial(SIP/xxx${extension}@media_gateway,10,t)
>>>
>>> but when I use:
>>>
>>> exten =>s,5,NoOp(SIP/xxx${extension}@media_gateway has state ${DIALSTATUS})
>>>        
>> I'm not quite sure what you are trying to do.
>>
>> So you called the phone for 10 seconds, the phone didn't answer - and
>> the variable "DIALSTATUS" told you exactly that.
>>
>> Is the problem the fact that the line is not ringing out? Is that what
>> is wrong?
>>
>> And why do you have some "xxx" in front of ${extension}? You shouldn't
>> need them. Just pass ${extension} - which is the number you dialled on
>> the phone.
>>
>> Sebastian
>>
>>
>>      
>>> I hear a busy tone, after the 10 sec. timeout it returns NOANSWER, as defined in my DIAL func.
>>>
>>> I also tried getting the DEVICE_STATE
>>>
>>> exten =>s,3,NoOp(SIP/xxx${extension}@media_gateway has state ${DEVICE_STATE(SIP/xxx${extension}@media_gateway)})
>>>
>>> and same thing happens as stated on the scenario below.
>>>
>>> Thanks again!
>>>
>>> regards,
>>>
>>> RYAN ICASIANO
>>> ________________________________________
>>> From: asterisk-users-bounces at lists.digium.com [asterisk-users-bounces at lists.digium.com] On Behalf Of Sebastian [shop at open-t.co.uk]
>>> Sent: Wednesday, October 27, 2010 5:00 PM
>>> To: asterisk-users at lists.digium.com
>>> Subject: Re: [asterisk-users] Mobile Phones and Asterisk
>>>
>>> Hi,
>>>
>>> On 10/27/2010 05:55 AM, GBR Icasiano, Ryan A. wrote:
>>>        
>>>> anyone???
>>>>
>>>> regards,
>>>>
>>>> RYAN ICASIANO
>>>>
>>>> Hi,
>>>>
>>>> I changed my sip.conf and added call-limit. At first I thought it works ok, since i tried calling a cellphone that is currently busy(phone answers 1st softphone, then another softphone calls the same number, it now returns INUSE). But then, i tried calling a different number while the first phone is busy, but it returns INUSE. It seems that the status being returned was from the peer itself(both phones uses the same peer) and not from the device(mobile phone) which i believe is more logical.
>>>>
>>>> I also tried using DIALSTATUS(which of course you need to DIAL first), but then I only hear a busy tone and the dialstatus will return a noanswer. Do I have to configure it first in order to capture the busy status of a device? Have you done something similar to this?
>>>>
>>>> I'm using ver. 1.6. Thanks in advance.
>>>>          
>>> I'm not sure I understand your setup. Are you using SIP for trunking, or
>>> for extensions? Are you calling a normal mobile phone, or a SIP client
>>> on a mobile phone?
>>>
>>> Sebastian
>>>
>>>        
>>>> regards,
>>>>
>>>> RYAN ICASIANO
>>>> ________________________________________
>>>> From: asterisk-users-bounces at lists.digium.com [asterisk-users-bounces at lists.digium.com] On Behalf Of GBR Icasiano, Ryan A. [raicasiano at globalbridgeresources.com]
>>>> Sent: Tuesday, October 26, 2010 10:41 AM
>>>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>>>> Subject: [asterisk-users] Mobile Phones and Asterisk
>>>>
>>>> Hi,
>>>>
>>>> Is the dev_state can also be used  to track a mobile phone's status via SIP? I tried it on several phones(nokia, samsung) but it returns NOANSWER but i can hear a beep beep beep sound indicating that it is currently busy.
>>>>
>>>> regards,
>>>>
>>>> RYAN ICASIANO
>>>>
>>>> __________________________
>>>> From: asterisk-users-bounces at lists.digium.com [asterisk-users-bounces at lists.digium.com] On Behalf Of Sebastian [shop at open-t.co.uk]
>>>> Sent: Tuesday, October 26, 2010 7:50 PM
>>>> To: asterisk-users at lists.digium.com
>>>> Subject: Re: [asterisk-users] Mobile Phones and Asterisk
>>>>
>>>> On 10/26/2010 12:30 PM, ayodele abejide wrote:
>>>>          
>>>>> Hello Jonathan,
>>>>>
>>>>> The solution would work only if the ISP has one public address, but in
>>>>> my solution they have a pool of public address, any other possible solution?
>>>>>            
>>>> With dynamic dns, you either install a piece of software on your server
>>>> (dynamic dns client) or you use the facility provided by your router
>>>> (some firewall/router/access point combo's have them). This software
>>>> updates automatically the record with dyndns every time your IP address
>>>> changes.
>>>>
>>>> Sebastian
>>>>
>>>>
>>>>          
>>>>> ABEJIDE, Ayodele A. (CCNA)
>>>>> +2348039269311
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> ------------------------------------------------------------------------
>>>>> From: ayodeleabejide at hotmail.com
>>>>> To: asterisk-users at lists.digium.com
>>>>> Date: Tue, 26 Oct 2010 11:01:09 +0000
>>>>> Subject: Re: [asterisk-users] Mobile Phones and Asterisk
>>>>>
>>>>> thanks i would check it up
>>>>>
>>>>> ABEJIDE, Ayodele A. (CCNA)
>>>>> +2348039269311
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> ------------------------------------------------------------------------
>>>>> Date: Tue, 26 Oct 2010 12:52:30 +0200
>>>>> From: jonathan.gsc at gmail.com
>>>>> To: asterisk-users at lists.digium.com
>>>>> Subject: Re: [asterisk-users] Mobile Phones and Asterisk
>>>>>
>>>>> Try http://www.dyndns.com/ that should solve your problem with dynamic IPs.
>>>>>
>>>>> Regards,
>>>>> Jonathan
>>>>>
>>>>> On Tue, Oct 26, 2010 at 12:40 PM, ayodele abejide
>>>>> <ayodeleabejide at hotmail.com<mailto:ayodeleabejide at hotmail.com>>     wrote:
>>>>>
>>>>>         Dear Asterisk-Users,
>>>>>
>>>>>         I have this Asterisk Box I run in my house, I need to terminate and
>>>>>         originate remote calls through the box via internet (SIP), the
>>>>>         problem is in Nigeria most ISPs would not provide you with Public
>>>>>         Addresses, all they provide is dynamic Natted addresses which change
>>>>>         each time one connects, I have thought of all possible solutions and
>>>>>         cannot come up with one, can anyone please help.
>>>>>
>>>>>         Thanks in anticipation
>>>>>
>>>>>         ABEJIDE, Ayodele A. (CCNA)
>>>>>         +2348039269311
>>>>>
>>>>>
>>>>>
>>>>>         --
>>>>>         _____________________________________________________________________
>>>>>         -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>>>>         New to Asterisk? Join us for a live introductory webinar every Thurs:
>>>>>         http://www.asterisk.org/hello
>>>>>
>>>>>         asterisk-users mailing list
>>>>>         To UNSUBSCRIBE or update options visit:
>>>>>         http://lists.digium.com/mailman/listinfo/asterisk-users
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> --
>>>>> Personal webpage - www.jonbaraq.eu<http://www.jonbaraq.eu>
>>>>>
>>>>> -- _____________________________________________________________________
>>>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>>>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>>>>> http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE
>>>>> or update options visit:
>>>>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>>>> -- _____________________________________________________________________
>>>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>>>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>>>>> http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE
>>>>> or update options visit:
>>>>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>>>>
>>>>>            
>>>> --
>>>> _____________________________________________________________________
>>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>>>>                    http://www.asterisk.org/hello
>>>>
>>>> asterisk-users mailing list
>>>> To UNSUBSCRIBE or update options visit:
>>>>        http://lists.digium.com/mailman/listinfo/asterisk-users
>>>>
>>>>          
>>> --
>>> _____________________________________________________________________
>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>>>                   http://www.asterisk.org/hello
>>>
>>> asterisk-users mailing list
>>> To UNSUBSCRIBE or update options visit:
>>>       http://lists.digium.com/mailman/listinfo/asterisk-users
>>>
>>>        
>> --
>> _____________________________________________________________________
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>>                  http://www.asterisk.org/hello
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>      http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>>      
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>                 http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>     http://lists.digium.com/mailman/listinfo/asterisk-users
>
>

More information about the asterisk-users mailing list