There is nothing special about the version of Asterisk that's in Astlinux that would affect SIP behavior. I would ask on the Asterisk mailing list.
Andrea Cristofanini wrote: > Dear list, > using astlinux i see this : > > When one extension make a call to another extension the remote party > display calling numer and called number . > In any other asterisk env i have that called party display only calling > number . > > Here the sip message : > > INVITE sip:1...@192.168.210.67:5060;rinstance=2b7d14b2e0f5cbf6 SIP/2.0 > Via: SIP/2.0/UDP 192.168.210.234:5060;branch=z9hG4bK32cc5a92;rport > *From: "113" <sip:1...@192.168.210.234>;tag=as35abae33* > To: <sip:1...@192.168.210.67:5060;rinstance=2b7d14b2e0f5cbf6> > *Contact: <sip:1...@192.168.210.234>* > Call-ID: 01a9b41b44bc67a2588ee338435fd...@192.168.210.234 > CSeq: 102 INVITE > User-Agent: Faro-PBX > Max-Forwards: 70 > Date: Tue, 12 May 2009 13:13:03 GMT > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Supported: replaces > Content-Type: application/sdp > Content-Length: 399 > > Anyone know where this coming from ? > > Obviusly my dialplan does not override any callerid. > I'm only using the sdtexten macro. > Regards ------------------------------------------------------------------------------ The NEW KODAK i700 Series Scanners deliver under ANY circumstances! Your production scanning environment may not be a perfect world - but thanks to Kodak, there's a perfect scanner to get the job done! With the NEW KODAK i700 Series Scanner you'll get full speed at 300 dpi even with all image processing features enabled. http://p.sf.net/sfu/kodak-com _______________________________________________ Astlinux-users mailing list Astlinux-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/astlinux-users Donations to support AstLinux are graciously accepted via PayPal to pay...@krisk.org.
