Displaying 8 results from an estimated 8 matches for "clientcode".
2010 Jul 13
1
Equivalent of SAS's FIRST. And LAST. Variable in R?
Hi all,
I'm just wondering if there is a equivalent of SAS's FIRST. and LAST.
variables in R?
For example, suppose this is a snapshot of the data:
ClientCode CaseCode open close Important
1 37 28 2003-07-08 2003-09-02 1
2 37 310 2003-11-01 2004-09-10 1
3 37 1562 2007-04-03 2007-07-27 1
4 38 29 2003-02-28 2007-09-05 1
5 38 599...
2010 Nov 06
2
One way voice with Asterisk
...ify hold state: No
SIP Transfer mode: open
Max Call Bitrate: 384 kbps
Auto-Framing: No
Default Settings:
-----------------
Context: default
Nat: RFC3581
DTMF: rfc2833
Qualify: 0
Use ClientCode: No
Progress inband: Never
Language: (Defaults to English)
MOH Interpret: default
MOH Suggest:
Voice Mail Extension: asterisk
----
Parsing /etc/asterisk/extconfig.conf
sip show peer
* Name : 155
Secret :<Set>
MD5Sec...
2017 Apr 04
0
AST-2017-001: Buffer overflow in CDR's set user
...This currently affects any system using CDR's that also
make use of the following:
* The 'X-ClientCode' header within a SIP INFO message when
using chan_sip and
the 'useclientcode' option is enabled (note, it's disabled...
2017 Nov 08
0
AST-2017-010: Buffer overflow in CDR's set user
...This currently affects any system using CDR's that also
make use of the following:
* The 'X-ClientCode' header within a SIP INFO message when
using chan_sip and
the 'useclientcode' option is enabled (note, it's disabled...
2011 Apr 18
2
Registrations stops after 403 FORBIDDEN
Hello list,
I have in sip.conf :
/maxexpiry=60 ; Maximum allowed time of incoming
registrations
; and subscriptions (seconds)
minexpiry=60 ; Minimum length of
registrations/subscriptions (default 60)
defaultexpiry=120 ; Default length of incoming/outgoing
registration
;-----------------------------------------
2006 Dec 18
0
pap2/wrt54gs/asterisk
...Reg. default duration: 120 secs
Outbound reg. timeout: 20 secs
Outbound reg. attempts: 0
Notify ringing state: Yes
Default Settings:
-----------------
Context: default
Nat: RFC3581
DTMF: rfc2833
Qualify: 0
Use ClientCode: No
Progress inband: Never
Language: (Defaults to English)
Musicclass: default
Voice Mail Extension: asterisk
******************sip.conf file*************************
GNU nano 1.3.8 File: sip.conf
[general]
c...
2011 May 08
3
Unable to REGISTER to the Asterisk v1.8.3.3 server via SIP/TLS
...r: uas
Session Expires: 1800 secs
Session Min-SE: 90 secs
Timer T1: 3000
Timer T1 minimum: 100
Timer B: 192000
No premature media: Yes
Max forwards: 70
Default Settings:
-----------------
Allowed transports: UDP
Outbound transport: UDP
Context: default
Force rport: No
DTMF: rfc2833
Qualify: 0
Use ClientCode: No
Progress inband: Never
Language:
MOH Interpret: default
MOH Suggest:
Voice Mail Extension: asterisk
*CLI> sip show peer 361
* Name : 361
Secret : <Set>
MD5Secret : <Not set>
Remote Secret: <Not set>
Context : family
Subscr.Cont. : <Not set>
Language :
AMA flags : Un...
2015 Mar 03
6
TLS, SRTP, Asterisk11 and Snom870s
CentOS-6.5 (FreePBX-2.6)
Asterisk-11.14.2 (FreePBX)
snom870-SIP 8.7.3.25.5
I am having a very difficult time attempting to get TLS and SRTP
working with Asterisk and anything else. At the moment I am trying to
get TLS functioning with our Snom870 desk-sets. And I am not having
much luck.
Since this is an extraordinarily (to me) Byzantine environemnt I am
going to ask if any of you have gotten