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${extensi...@media_gateway,10,t)
but when I use:
exten =>s,5,NoOp(SIP/xxx${extensi...@media_gateway has state ${DIALSTATUS})
I hear a busy tone, after the 10 sec. timeout it returns NOANSWER, as defined
in my DIAL func.
I also tried getting the DEVICE_STATE
exten =>s,3,NoOp(SIP/xxx${extensi...@media_gateway has state
${DEVICE_STATE(SIP/xxx${extensi...@media_gateway)})
and same thing happens as stated on the scenario below.
Thanks again!
regards,
RYAN ICASIANO
________________________________________
From: [email protected]
[[email protected]] On Behalf Of Sebastian
[[email protected]]
Sent: Wednesday, October 27, 2010 5:00 PM
To: [email protected]
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: [email protected]
> [[email protected]] On Behalf Of GBR Icasiano, Ryan A.
> [[email protected]]
> 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: [email protected]
> [[email protected]] On Behalf Of Sebastian
> [[email protected]]
> Sent: Tuesday, October 26, 2010 7:50 PM
> To: [email protected]
> 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: [email protected]
>> To: [email protected]
>> 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: [email protected]
>> To: [email protected]
>> 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
>> <[email protected]<mailto:[email protected]>> 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
>>
>>
>>
>> --
>> _____________________________________________________________________
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>>
>>
>>
>> --
>> Personal webpage - www.jonbaraq.eu<http://www.jonbaraq.eu>
>>
>> -- _____________________________________________________________________
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>> New to Asterisk? Join us for a live introductory webinar every Thurs:
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>
> --
> _____________________________________________________________________
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> New to Asterisk? Join us for a live introductory webinar every Thurs:
> http://www.asterisk.org/hello
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