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Joe-Palmer / rtmplite

Automatically exported from code.google.com/p/rtmplite
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Cannot dial sip address with * in front #15

Closed GoogleCodeExporter closed 9 years ago

GoogleCodeExporter commented 9 years ago 
Reported by Franck <infos@madovsky.org> on Aug 3, 2010

there is this message error when I try to call a sip address with an asterisk 
in front :

rtmp-invite  *010600@ekiga.net
exception in invite (<type 'exceptions.AttributeError'>, 
AttributeError("'NoneType' object has no attribute 'dup'",), <traceback object 
at 0x1f92b00>)
Gateway[0] onResult None

tried to correct it but my python level is not yet skillfull

Thanks

Franck

Original issue reported on code.google.com by voiprese...@gmail.com on 5 Feb 2011 at 12:18

GoogleCodeExporter commented 9 years ago 
The correct syntax is to use a SIP URL, e.g., sip:*010600@ekiga.net, which 
seems to work in sending out an INVITE, but fails due to "488 Not Acceptable 
Here" because siprtmp/Flash Player only sends Speex.

I have added error checking in p2p-sip's svn r23 to return 'invalid dest URI' 
error if URI is invalid and cannot be parsed correctly.

Original comment by voiprese...@gmail.com on 5 Feb 2011 at 2:09