[Asterisk-Users] Calling an outside number along side otherinternal extensions?

[Paul Fielding]

Hmmm... Interesting that you mention it's not a problem with VOIP companies 
as they use PRI.  The analog trunk I'm connecting to is actually a Vonage 
line.  Thing is, it seems to me that by connecting via the Zap channel to 
the Vonage ATA I'm effectively cancelling any advantage that Vonage's PRI 
might have... (?).    I don't believe I have any other alternatives for 
connecting to Vonage's service, but perhaps I'm wrong about that.

Perhaps I'll give the "c" option a try.  It looks like it might do the 
job...

regards,

Paul

----- Original Message ----- 
From: "Eric Wieling" <eric at fnords.org>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 
<asterisk-users at lists.digium.com>
Sent: Friday, November 12, 2004 6:56 PM
Subject: Re: [Asterisk-Users] Calling an outside number along side 
otherinternal extensions?


> Paul Fielding wrote:
>>
>>
>> I've currently configured incoming calls to simultaneously ring an analog 
>> phone (via TDM400P) and two SIP phones.   I'd like to have it also 
>> simultaneously dial out the TDM400P on a PSTN to ring my cell phone, and 
>> have the first one to answer win the battle.
>>
>>  In my digging I've figured out that I can add the Zap channel to the 
>> dial list, such as Dial(SIP/7001&SIP/7002&ZAP/3/5551212,20), however when 
>> I include the PSTN line in this command (ZAP/3/....) I get an interesting 
>> thing happening.
>>
>>  All SIP phones start ringing.
>>
>> Asterisk connects ZAP/3 to dial out and dials out
>>
>> Asterisk then says to the effect of "ZAP/3 has answered the call" (since 
>> the line has now gone off hook) and stops ringing all the SIP phones 
>> immediately, leaving me with only the cell phone ringing.  It then fails 
>> to go to Voicemail and just keeps ringing the cell phone, because as far 
>> as Asterisk is concerned the line has been bridged and is connected.
>>
>>  Any suggestions?
>
> Analog FXO ports ae considered "answered" as soon as the dialing is 
> finished.  Nothing you can do about this because there is no way for 
> Asterisk to know when the far end answers.  This is not a problem with 
> (most) Channelized Voice T-1, it's not a problem with PRI and not a 
> problem with VoIP telephone companies, since they all use PRI.
>
> You can sort of work around this problem by using the poorly documented 
> "c" option to the Zap dial command.  Something like Zap/1c or something 
> like that.  I've never used it.  That option requires the callee press # 
> to accept the call.  No sound file is played.  See the mailing list 
> archives.  It's been discussed off and on.
>
> --Eric
> --Eric
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