[Asterisk-Dev] variable issue in app_dial 1.102

[Michalis Manousos]

On Thu, 4 Nov 2004, Asterisk mailing list box wrote:


The new dial app does not copy to the new channel created by it just 
some special variables (like the ALERT_INFO). It copies channel
variables based on their name. If the first character of the variable's
name is '_' then the variable is copied to the channel and the initial
underscore is removed (so, a second dial won't pass the variable). If
the variable's name start with '__' (two underscores) then the variable
is copied to the new channel without removing the underscores (so,
additional dial()s will always copy this variable. If the variable's name
doesn't start with underscore, the variable is not copied.

For your case, set an _ALERT_INO variable and it will work.

Michael.


> 
> I cleared out everything and now have CVS-HEAD-11/04/04-12:20:25 and
> it still fails with the new code but works with the older code.
> 
> Anyone have any thoughts or see something wrong?
> 
> I set "sip debug peer 204.213.176.201" for the receiving phone.
> 
> On the "old" code I get a correct header:
> Reliably Transmitting:
> INVITE sip:301f at 204.213.176.201 SIP/2.0
> Via: SIP/2.0/UDP 204.213.176.174:5060;branch=z9hG4bK05fb41eb;rport
> From: "Test Polycom" <sip:381 at 204.213.176.174>;tag=as4b17304d
> To: <sip:301f at 204.213.176.201>
> Contact: <sip:381 at 204.213.176.174>
> Call-ID: 256816892b6069283e4323a130d8d77d at 204.213.176.174
> CSeq: 102 INVITE
> User-Agent: Asterisk PBX
> Date: Thu, 04 Nov 2004 17:33:11 GMT
> Alert-info: ICM
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
> Content-Type: application/sdp
> Content-Length: 293
> 
> With the current "new" code. The Alert-info is missing:
> Reliably Transmitting:
> INVITE sip:301f at 204.213.176.201 SIP/2.0
> Via: SIP/2.0/UDP 204.213.176.174:5060;branch=z9hG4bK694a054f;rport
> From: "Test Polycom" <sip:381 at 204.213.176.174>;tag=as55b62178
> To: <sip:301f at 204.213.176.201>
> Contact: <sip:381 at 204.213.176.174>
> Call-ID: 7befb2727d43b5ff62cf397545695ff4 at 204.213.176.174
> CSeq: 102 INVITE
> User-Agent: Asterisk PBX
> Date: Thu, 04 Nov 2004 17:27:59 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
> Content-Type: application/sdp
> Content-Length: 293
> 
> 
> The extensions.conf file is the same. Only the app_dial.c is different.
> 
> [intercom]
> exten => _63XX,1,SetVar(ALERT_INFO=ICM)
> exten => _63XX,n,Dial(SIP/${EXTEN:1}f,5,m)
> exten => _63XX,n,Playback(intercombusy)
> exten => _63XX,n,Hangup()
> 
> 
> >Date: Wed, 3 Nov 2004 15:52:31 -0500
> >From: Matthew Marlowe <matthew.marlowe at gmail.com>
> >To: Asterisk mailing list box <asterisk at ntplx.net>, Asterisk Developers Mailing 
> List <asterisk-dev at lists.digium.com>
> >Subject: Re: [Asterisk-Dev] variable issue in app_dial 1.102
> >
> >I'm running CVS-HEAD-11/03/04-14:09:34 and it works for me.
> >
> >
> >On Mon, 1 Nov 2004 16:57:24 -0500 (EST), Asterisk mailing list box
> ><asterisk at ntplx.net> wrote:
> >> 
> >> app_dial version 1.99 works for me. On the CVS update to 1.102 the
> >> ALERT_INFO variable I set no longer works. Everything else is up to date.
> >> If I back off to the older app_dial it works again.
> >> 
> >> I have an extension that includes SetVar(ALERT_INFO=ICM) which I use
> >> to set auto answer for a intercom line on Polycom phones. It worked before.
> >> 
> >> Does the variable need to be set someother way now? I notice the code in
> >> app_dial.c is no longer looking for ALERT_INFO....
> >> 
> >>  Andrew
> >> 
> >> _______________________________________________
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> >> 
> >
> >
> >-- 
> >MBM
> 
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