[Voiceglue] Voiceglue Digest, Vol 29, Issue 3

Mon Dec 7 16:51:50 UTC 2009

I was playing with different options available. The dial mode keeps a vxml dialog open, which in turn allows us to track conversation time easily. The first option "transfer" will work but requires more work to do that...

> From: voiceglue-request at voiceglue.org
> Subject: Voiceglue Digest, Vol 29, Issue 3
> To: voiceglue at voiceglue.org
> Date: Mon, 7 Dec 2009 12:00:01 +0000
> 
> Send Voiceglue mailing list submissions to
> 	voiceglue at voiceglue.org
> 
> To subscribe or unsubscribe via the World Wide Web, visit
> 	http://www.voiceglue.org/mailman/listinfo/voiceglue
> or, via email, send a message with subject or body 'help' to
> 	voiceglue-request at voiceglue.org
> 
> You can reach the person managing the list at
> 	voiceglue-owner at voiceglue.org
> 
> When replying, please edit your Subject line so it is more specific
> than "Re: Contents of Voiceglue digest..."
> 
> 
> Today's Topics:
> 
>    1. transfer tag problem with the dial option (Denis Orlov)
>    2. transfer tag problem with the dial option (Denis Orlov)
>    3. Re: transfer tag problem with the dial option (Doug Campbell)
> 
> 
> ----------------------------------------------------------------------
> 
> Message: 1
> Date: Sun, 6 Dec 2009 11:36:47 -0800
> From: "Denis Orlov" <denrl at hotmail.com>
> To: <voiceglue at voiceglue.org>
> Subject: [Voiceglue] transfer tag problem with the dial option
> Message-ID: <SNT129-DS18CD417173078FE342D84ABE910 at phx.gbl>
> Content-Type: text/plain; charset="iso-8859-1"
> 
> Hello,
> 
> I am using the transfer tag. It works fine with the default blind_xfer_method=transfer. However, when set to "dial", I get the following exception from Asterisk:
> 
> [Dec  6 11:19:34] VERBOSE[8239] logger.c:     -- AGI Script Executing Application: (Dial) Options: (sip/1002||)
> [Dec  6 11:19:34] VERBOSE[8239] logger.c:   == Using SIP RTP TOS bits 184
> [Dec  6 11:19:34] VERBOSE[8239] logger.c:   == Using SIP RTP CoS mark 5
> [Dec  6 11:19:35] WARNING[8239] chan_sip.c: No such host: 1002||
> [Dec  6 11:19:35] WARNING[8239] app_dial.c: Unable to create channel of type 'sip' (cause 20 - Unknown)
> 
> Apparently, system adds "||" characters to the end, and it causes the problem with the host resolution...
> 
> Is it a bug or I am doing something wrong? 
> 
> I use 0.11 on Linux Fedora 9.
> 
> Thanks
> 
> Denis
> 
> -------------- next part --------------
> An HTML attachment was scrubbed...
> URL: <http://www.voiceglue.org/pipermail/voiceglue/attachments/20091206/97d2d2a4/attachment-0001.html>
> 
> ------------------------------
> 
> Message: 2
> Date: Sun, 6 Dec 2009 11:44:44 -0800
> From: "Denis Orlov" <denrl at hotmail.com>
> To: <voiceglue at voiceglue.org>
> Subject: [Voiceglue] transfer tag problem with the dial option
> Message-ID: <SNT129-DS20BD58CFFA42D9CD8BC3F9BE910 at phx.gbl>
> Content-Type: text/plain; charset="iso-8859-1"
> 
> Hello,
> 
> I am using the transfer tag. It works fine with the default blind_xfer_method=transfer. However, when set to "dial", I get the following exception from Asterisk:
> 
> [Dec  6 11:19:34] VERBOSE[8239] logger.c:     -- AGI Script Executing Application: (Dial) Options: (sip/1002||)
> [Dec  6 11:19:34] VERBOSE[8239] logger.c:   == Using SIP RTP TOS bits 184
> [Dec  6 11:19:34] VERBOSE[8239] logger.c:   == Using SIP RTP CoS mark 5
> [Dec  6 11:19:35] WARNING[8239] chan_sip.c: No such host: 1002||
> [Dec  6 11:19:35] WARNING[8239] app_dial.c: Unable to create channel of type 'sip' (cause 20 - Unknown)
> 
> Apparently, system adds "||" characters to the end, and it causes the problem with the host resolution...
> 
> Is it a bug or I am doing something wrong? 
> 
> I use 0.11 on Linux Fedora 9.
> 
> Thanks
> 
> Denis
> 
> -------------- next part --------------
> An HTML attachment was scrubbed...
> URL: <http://www.voiceglue.org/pipermail/voiceglue/attachments/20091206/3d47a4aa/attachment-0001.html>
> 
> ------------------------------
> 
> Message: 3
> Date: Sun, 06 Dec 2009 18:09:38 -0500
> From: Doug Campbell <voiceglue at campbellcastle.com>
> To: General discussion about voiceglue <voiceglue at voiceglue.org>
> Subject: Re: [Voiceglue] transfer tag problem with the dial option
> Message-ID: <4B1C39B2.5020601 at campbellcastle.com>
> Content-Type: text/plain; charset=ISO-8859-1; format=flowed
> 
> > I am using the transfer tag. It works fine with the default 
> > blind_xfer_method=transfer. However, when set to "dial", I get the 
> > following exception from Asterisk:
> 
> Be grateful you are getting at least one of the ways to work!  :-)
> The transfer functionality has always been a second-class citizen
> in voiceglue.  I suppose it would be nice to get it cleaned up
> properly at some point.
> 
> Out of curiosity, what is it about the working method that is
> unacceptable to you that thus requires the "dial" method?
> 
> Thanks,
> Doug Campbell
> 
> 
> ------------------------------
> 
> _______________________________________________
> Voiceglue mailing list
> Voiceglue at voiceglue.org
> http://www.voiceglue.org/mailman/listinfo/voiceglue
> 
> 
> End of Voiceglue Digest, Vol 29, Issue 3
> ****************************************

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://www.voiceglue.org/pipermail/voiceglue/attachments/20091207/8014528d/attachment.html>