[asterisk-bugs] [Asterisk 0017285]: SIP Anonymous Call: h323 to SIP
Asterisk Bug Tracker
noreply at bugs.digium.com
Wed May 5 10:58:26 CDT 2010
The following issue has been CLOSED
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https://issues.asterisk.org/view.php?id=17285
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Reported By: Steph63
Assigned To:
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Project: Asterisk
Issue ID: 17285
Category: Channels/chan_h323
Reproducibility: always
Severity: minor
Priority: normal
Status: closed
Asterisk Version: Addons-1.6.2.0
JIRA:
Regression: No
Reviewboard Link:
SVN Branch (only for SVN checkouts, not tarball releases): N/A
SVN Revision (number only!):
Request Review:
Resolution: open
Fixed in Version:
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Date Submitted: 2010-05-04 18:13 CDT
Last Modified: 2010-05-05 10:58 CDT
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Summary: SIP Anonymous Call: h323 to SIP
Description:
Hi,
I am an Asterisk newbie and I think that there might be something wrong
about the way the h323 ID is transfer from an H323 to SIP call.
I am expecting some behaviour like OOH323, may be this is where I am
wrong, where a call from NetMeeting or Ekiga (h323) is displayed on X-lite
with the H323 ID defined in the First Name/Last Name of NetMeeting
Directory information or in Ekiga first Name and Surname information.
The problem with h323 is that calls originated from an h323 soft phone are
not match with any h323id defined in h323. The SIP phones displays a call
from Anonymous SIP call whether the h323 id is defined or not in h323.conf
Found similiar behavior for
[root at samba3 asterisk]# cat /etc/redhat-release
CentOS release 5.4 (Final)
samba3*CLI> show version
Asterisk 1.6.2.0 built by root @ samba2.maison.net on a i686 running Linux
on 2010-03-20 23:51:46 UTC
or
[root at samba3 asterisk]# cat /etc/redhat-release
CentOS release 5.2 (Final)
samba3*CLI> show version
Asterisk 1.4.30 built by root @ samba3.maison.net on a i686 running Linux
on 2010-04-20 23:51:46 UTC
and
Here is my configuration
h323.conf
[general]
port = 1720
bindaddr = 0.0.0.0
disallow=all
allow=ulaw
dtmfmode=inband
gatekeeper = DISABLE
context=default
[Bob Smith]
type=user
host=192.168.1.100
context=phones
sip.conf
[general]
context=phones
bindport=5060
bindaddr=0.0.0.0
[1000]
type=friend
username=1000
secret=farinon
host=dynamic
nat=no
context=phones
disallow=all
allow=ulaw
allow=h263
allow=h263p
canreinvite=yes
mailbox=1100 at phones
extensions.conf
[phones]
exten => 1000,1,Dial(SIP/1000,15)
exten => 1000,2,VoiceMail(1100 at phones)
exten => 1000,3,PlayBack(vm-goodbye)
exten => 1000,4,Wait(2)
exten => 1000,5,HangUp()
exten => 4000,1,Dial(h323/192.168.1.100,100,Ttr)
Debug
samba3*C== New H.323 Connection created.
samba3*C--Received SETUP message
-- Setting up Call
-- LI> Call token: [ip$192.168.1.100:1538/23325]
-- LI> Calling party name: [Bob Smith]
-- LI> Calling party number: []
-- LI> Called party name: [1000]
-- LI> Called party number: [1000]
-- LI> Calling party IP: [192.168.1.100]
Setting capabilities to 0x4 (ulaw)
Capabilities in preference order is (ulaw)
Allowed Codecs:
samba3*C Table:
G.711-uLaw-64k <1>
UserInput/hookflash <2>
UserInput/basicString <3>
Set:3*CLI>
0:3*CLI>
0:CLI>
G.711-uLaw-64k <1>
1:CLI>
UserInput/hookflash <2>
2:CLI>
UserInput/basicString <3>
samba3*CLI>
samba3*C=-= In OnAnswerCall for call 23325
samba3*CLI> - Progress Indicator: 0
samba3*CLI> - Inserting PI of 0 into ALERTING message
[Apr 25 21:11:50] NOTICE[6561]: chan_h323.c:2229 answer_call: Dropping
call because extensions '1000', 's' and 'i' doesn't exists in context
[default]
samba3*C-- Sending RELEASE COMPLETE
samba3*C-- ClearCall: Request to clear call with token
ip$192.168.1.100:1538/23325, cause EndedByNoUser
samba3*C-- ClearCall: Request to clear call with token
ip$192.168.1.100:1538/23325, cause EndedByAnswerDenied
samba3*C-- ClearCall: Request to clear call with token
ip$192.168.1.100:1538/23325, cause EndedByTransportFail
-- Remote endpoint could not find user: Bob Smith [192.168.1.100]
samba3*C== H.323 Connection deleted.
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Issue History
Date Modified Username Field Change
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2010-05-05 10:58 pabelanger Status new => closed
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