[asterisk-users] Mobile Phones and Asterisk

GBR Icasiano, Ryan A. raicasiano at globalbridgeresources.com
Wed Oct 27 19:06:13 CDT 2010


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 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



More information about the asterisk-users mailing list