Re: [asterisk-users] Connect mobile to asterisk

2010-05-31 Thread lesouvage
If you are interested in really integrating GSM phones into an  
Asterisk based system without any telco involved check the OpenBTS  
project. I have done a research and trial project and this combination  
of open hardware (USRP), the OpenBTS open source project and Asterisk  
is pretty amazing. The hardware is kind of expensive and you have to  
check the legislation in the country if you plan to use an OpenBTS  
based solution but the whole project is real interesting.

  Erik


On 31 mei 2010, at 12:12, Vincent wrote:

> On Sat, 29 May 2010 11:31:05 +0530 (IST), Nivin Kumar
>  wrote:
>> I would like to connect my blackberry or any other cell phone to  
>> asterisk so that
>> I can send calls through the sim card. I would also like to send  
>> SMS through this as well.
>
> Since wifi isn't as reliable and pervasive as GSM (and I read that
> BlackBerry don't allow VoIP clients anyway, so as to force users to
> make calls through their cellphone provider), I assume you don't want
> to connect the Blackberry to Asterisk through either through
> USB/Ethernet or wifi, but rather through GSM.
>
> The only solution I know is to buy a GSM gateway that will be
> connected by wire to your Asterisk server at home, and you'll need to
> get a second GSM subscription so that the GSM gateway has a SIM and
> let you connect your GSM phone to your Asterisk server.
>
> www.voip-info.org/wiki/view/VOIP+GSM+Gateways
>
> If someone knows of a cheaper way to connect a GSM phone to an
> Asterisk server, I'm also interested.
>
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Re: [asterisk-users] Continuing after a TIMEOUT(absolute)

2010-05-13 Thread lesouvage
The whole idea of TIMEOUT(absolute) is to end to call after a certain  
time. My advice is to explain what you are trying to achieve, there  
might be a solutions but I doubt you will find it while using  
TIMEOUT(absolute). If the dial plan reaches the t or T extension there  
are, as far as I know,  no (bridged) legs of a call left, just some  
channel variables you can use.


Erik
On 30 apr 2010, at 18:31, Brendan Sterne wrote:

> CF,
>
> When I comment out the timeout the call continues as expected.  I
> believe the timeout is kicking in.
>
> Can anyone point me to an example where TIMEOUT(absolute) is used as a
> general timer, where the call continues after the expiry?  I'm not
> sure which extension to use "T" or "t".  I've tried both but neither
> seem to work.
>
> Cheers,
> - Brendan
>
> Brendan Sterne
> QA Lead, Callvine
>
>
>
> On Apr 30, 2010, at 9:38 AM, C F wrote:
>
>> I don't think you are actually hitting the time out. Comment out the
>> set timeout line I think the results will be the same. Which tells me
>> the timeout is not kicking in.
>>
>> On 4/29/10, Brendan Sterne  wrote:
>>> Greetings,
>>>
>>> I'm trying to continue to do some processing after a TIMEOUT
>>> (absolute).  In my dialplan below, when a call comes in to  
>>> [default],
>>> I call macro-phonenum and pass it a timeout of 20 seconds.  macro-
>>> phonenum sets TIMEOUT(absolute), then loops saying the phone number
>>> that was called (in MACRO_EXTEN).  When the timeout expires I want  
>>> to
>>> call my macro-hangup (so it can say "goodbye" or whatever).  But the
>>> system is just hanging up.  The dialplan and log output is below.
>>> Any
>>> info is appreciated.  This is on version 1.6.0.5.
>>>
>>>
>>>
>>> [macro-answer-and-join]
>>> exten => s,1,NoOp()
>>> exten => s,n,Answer()
>>> exten => s,n,Wait(4)
>>> exten => s,n,SendDTMF(1)
>>> exten => s,n,Wait(1)
>>> exten => s,n,SendDTMF(1)
>>> exten => s,n,MacroExit
>>>
>>> [macro-hangup]
>>> exten => s,1,NoOp()
>>> exten => s,n,Playback(goodbye)
>>> exten => s,n,Hangup()
>>> ;
>>> exten => T,1,NoOp()
>>> exten => T,n,Playback(goodbye)
>>> exten => T,n,Hangup()
>>>
>>> [macro-phonenum]
>>> exten => s,1,NoOp()
>>> exten => s,n,Macro(answer-and-join)
>>> exten => s,n,Set(TIMEOUT(absolute)=${ARG1})
>>> exten => s,n,Set(i=1000)
>>> exten => s,n,While($[${i} >= 1])
>>> exten =>  s,n,SayDigits(${MACRO_EXTEN})
>>> exten =>  s,n,Wait(5)
>>> exten =>  s,n,Set(i=$[${i} - 1])
>>> exten => s,n,EndWhile()
>>> exten => s,n,MacroExit
>>> ;
>>> exten => T,1,NoOp()
>>> exten => T,n,Macro(hangup)
>>> exten => T,n,MacroExit
>>>
>>>
>>> [default]
>>> exten => _X.,1,NoOp()
>>> exten => _X.,n,Macro(phonenum,20)
>>> exten => _X.,n,Macro(hangup)
>>> ;
>>> exten => T,1,NoOp()
>>> exten => T,n,Macro(hangup)
>>>
>>>
>>>
>>> The log when the timeout occurs:
>>>
>>>  (I'm in macro-phonenum)
>>>   --  Playing 'digits/5.ulaw' (language
>>> 'en')
>>>--  Playing 'digits/1.ulaw' (language
>>> 'en')
>>>--  Playing 'digits/2.ulaw' (language
>>> 'en')
>>>--  Playing 'digits/1.ulaw' (language
>>> 'en')
>>>--  Playing 'digits/2.ulaw' (language
>>> 'en')
>>>-- Executing [...@macro-phonenum:7] Wait("SIP/
>>> 70.124.61.17-082a69a8", "5") in new stack
>>>  == Spawn extension (macro-phonenum, s, 7) exited non-zero on 'SIP/
>>> 70.124.61.17-082a69a8' in macro 'phonenum'
>>>  == Spawn extension (macro-phonenum, s, 7) exited non-zero on 'SIP/
>>> 70.124.61.17-082a69a8'
>>> Scheduling destruction of SIP dialog 'D8FE9724-1DD1-11B2-9F1A-
>>> a4ef9db84...@192.168.1.98' in 32000 ms (Method: ACK)
>>> set_destination: Parsing  for address/port to
>>> send to
>>> set_destination: set destination to 70.124.61.17, port 5060
>>> Reliably Transmitting (NAT) to 70.124.61.17:5060:
>>> BYE sip:70.124.61.17:5060 SIP/2.0
>>> 
>>>
>>>
>>>
>>> Cheers,
>>> - Brendan
>>>
>>> Brendan Sterne
>>> QA Lead, Callvine
>>>
>>>
>>>
>>>
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>>
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>
>
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Re: [asterisk-users] help!!! Internal extensions not connect

2010-03-01 Thread lesouvage
You doesn't seem to have a proper context,extension,priority available  
for internal calls while you have one for outbound calls. To get more  
detailed help an even an  answer you have to provide more info. The  
cli output while trying to setup an internal call will help.

Erik

On 2 mrt 2010, at 00:31, carem gyssell nieto wrote:

> I have a problem with my internal extensions, I'm using Asterisk  
> 1.6.2.5 and freePBX 2.6. When I call betwen extensions these don't  
> connect. There is a long silence and finally hang up.
> I have an E1 whit r2 and I use openr2, but I don't have problems to  
> do calls  to the PSTN..my problem it only with internal extension.
>
> Please help...it's an Asterisk bug??.
>
> Thanks :S
>
> -- 
> Carem Gyssell Nieto Garcia
> -- 
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