Re: [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" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 
<[EMAIL PROTECTED]>
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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