[Asterisk-Users] Zap/g0/ to a Telstra Mobile

[Eric Wieling]

Shane Dalgleish wrote:

>  
> 
> 
>>-----Original Message-----
>>From: asterisk-users-bounces at lists.digium.com 
>>[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of 
>>Eric Wieling
>>Sent: Friday, 18 February 2005 2:34 AM
>>To: Asterisk Users Mailing List - Non-Commercial Discussion
>>Subject: Re: [Asterisk-Users] Zap/g0/ to a Telstra Mobile
>>
>>Howard Lowndes wrote:
>>
>>
>>>On Thu, 2005-02-17 at 15:51, asterisk at tragicflirt.com wrote:
>>>
>>>
>>>>I've installed a TDM400. Having a go with AMP.
>>>>
>>>>I would like incoming calls to be put throuhg to an 
>>
>>extension (at my 
>>
>>>>desk) and a mobile (cell phone) at the same time. Whichever 
>>
>>picks up, 
>>
>>>>gets the call..
>>>>
>>>>This should be possible with AMP (call groups, 
>>
>>200|201|0*0408xxxxxx), 
>>
>>>>but it didn't work, so I have created a custom-incoming in 
>>>>extensions-custom.conf
>>>>
>>>>What is happening is, The extension rings for about 5 secs 
>>
>>(as long as 
>>
>>>>it takes the TDM400 to dial the mobile number), then just 
>>
>>the telstra 
>>
>>>>mobile rings..
>>>>
>>>>
>>>>>From asterisk -vvvvvvvvvvvr
>>>>
>>>>   -- Goto (custom-incoming,s,1)
>>>>   -- Executing Dial("SIP/202-b424", 
>>>>"Zap/g0/0408xxxxxx&Sip/200|30|t") in new stack
>>>>   -- Called g0/0408xxxxxx
>>>>   -- Called 200
>>>>   -- SIP/200-fece is ringing
>>>>   -- SIP/200-fece is ringing
>>>>   -- SIP/200-fece is ringing
>>>>   -- SIP/200-fece is ringing
>>>>   -- Zap/2-1 answered SIP/202-b424
>>>
>>>
>>>This tend to indicate to me that the mobile system has 
>>
>>picked up the 
>>
>>>call request on the zap channel and that * therefore thinks 
>>
>>that the 
>>
>>>zap channel has picked up the call and will then bridge the zap 
>>>channel to the sip 202 channel and kill off the ringing on 
>>
>>the sip 200 channel.
>>
>>>I don't know that there is much you can do about this as 
>>
>>basically you 
>>
>>>are trying to get interaction on two different systems.
>>
>>No.  Analog ports are always considered "ANSWERED" as soon as 
>>Asterisk finishes dialing.  This is covered over and over and 
>>over again in the mailing list archives.  There are a few 
>>very ugly hacks to work around the problem.
>>
> 
> 
> Thanks Howard and Eric,
> 
> I did have a look around for this before I posted and I found a few
> references to:
> callprogress=yes   (in zapata.conf)
> 
> But also read that this only (kinda) works in the US.
> 
> Also had a brief look at BackgroundDetect, but it looks a bit rough
> 
> 
> 
> What I do need to do urgently however is get rid of the 5 or so seconds of
> silence and static noise between the time Zap says the call is answered and
> Telstra establishes the call and starts ringing again..
> 
> So what I'm thinking is perhaps:-
> 
> - Call the phones in the office
> - Call the mobiles seperately but at the same time
> - wait for a DMTF tone from the mobile (I think I could put up with that)
> - bridges the call to the mobile
> - But if a Sip phone answers the call first hangup on the mobile
> - bridges the call to the Sip phone
> 
> Any thoughts on how I would go about that?

You can replace the "t" option with "tr" at the end of your Dial line. 
  However, if the destination is busy then you may hear a couple of 
rings and then a busy sound.

Why are you using "t" in the first place?

You really do need a PRI or VoIP service provider.  These things are 
not really issues with PRI or VoIP.