1 msg[Asterisk 1.2 + TDM FXO] Incoming call not dete...
3 msgMedia gateways and video
1 msgservice provider connection problem
3 msgpickup application failed
3 msgMulti-SPAN (4xE1) Zap Group (Outbound)
26 msgFWD and IPCall
22 msgHow to check if a SIP phone is forwarded withou...
2 msgextension.conf with mysql
2 msgStrange migration problems from asterisk 1.2.13...

Presentation Restricted h.323-SIP issue
\ Lucian Gheorghe (7 Jan 2008)

14 msgno outgoing calls with Digium B410P
3 msgIncrease Volume - SIP
4 msgHPEC
5 msgzaptel programming
6 msgChange Default Voicemail Message
1 msgAsterisk High Availability and Clustering
18 msgWhich IP Phone is really the best?
1 msgNew site for feature wish-list: Asteriskideas.org
2 msg[FreeBSD 6.2] Error compiling Zaptel from Ports?
1 msgZap with SIP
Subject:Presentation Restricted h.323-SIP issue
Group:Asterisk-users
From:Lucian Gheorghe
Date:7 Jan 2008


 


Hi All,



I have a problem with incoming calls to Asterisk that go to SIP phones, when
the h.323 message contains:



.... ..11 = Screening indicator: Network-provided (0x03)

.01. .... = Presentation indicator: Presentation restricted (0x01)



for the calling party number, the Contact URI still contains the
<calling_party_number@xyz> when it should contain <anonymous@xyz> (or to
use another sip method for CLIR). I searched the internet and the docs, but
I couldn't find any method to change this behaveour. Is there any way to set
this (maybe from the dialplan) except for me having to re-write the
interworking bits of the code ?



Thanks,

Lucian



_______________________________________________
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users


© 2004-2008 readlist.com