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« on: 24 Jan 2014, 00:06:15 »
Op een andere ziggo lijn zelfe probleem.
Heb Alle instellingen nog eens nagelopen. Maar zie zo snel niets verkeerd staan ?
Excuses voor de lange post.
Network Status:
NAT Type :Port Restricted Cone NAT (STUN)
General Settings Account 1
SIP Server : sip.cheapconnect.net
SIP User ID : (SIP Gebruikersnaam uit mail Cheapconnect)
Authenticate ID : (SIP Gebruikersnaam uit mail Cheapconnect)
Authenticate Password : (Wachtwoord uit mail Cheapconnect)
Voice Mail UserID : 1233
Name : (Telefoonnummer)
Tel URI : Disable
Network Settings Account 1
Outbound Proxy : sip.cheapconnect.net
Secondary Outbound Proxy :
DNS Mode : A record
NAT Traversal : Keep-alive
Proxy-Require :
SIP Settings Account 1
SIP Registration : Yes
Unregister On Reboot : No
Register Expiration (m) : 60
Wait Time Retry Registration (s) : 20
Local SIP Port : 5062
SUBSCRIBE for MWI : No
Session Expiration (s) : 180
Min-SE (s) : 90
UAC Specify Refresher : Omit
UAS Specify Refresher : UAC
Force INVITE : No
Caller Request Timer : No
Callee Request Timer : No
Force Timer : No
Enable 100rel : No
SIP Transport : UDP
Symmetric RTP : No
Support SIP Instance ID : Yes
Validate Incoming SIP Messages : No
Check SIP User ID for Incoming INVITE : No
Authenticate Incoming INVITE : No
Only Accept SIP Requests from Known Servers : No
SIP T1 Timeout : 0,5 Sec
SIP T2 Interval : 4 Sec
Remove OBP from route : No
Codec Settings Account 1
DTMF : RFC2833
DTMF Payload Type : 101
Preferred Vocoder : G722, G729A/B
Preferred Video Codec : H263, H263+
Enable RFC5168 support : No
H.264 Image Size : QVGA
H.264 Payload Type : 99
H.263+ Payload Type : 100
L16-256 Payload Type : 98
H.263 Encoder Resolution : CIF
SRTP Mode : Disable
Silence Suppression : No
Voice Frames Per TX : 2
G723 Rate : 5.3kpbs encoding rate
Jitter Buffer Type : Adaptive
Jitter Buffer Length : Medium
Send Silent RTP Packets on Mute : No
Call Settings Account 1
Start Video Automatically : Yes
Remote Video Request : Prompt
Dial Plan Prefix :
DialPlan : { x+ | \+x+ | *x+ | *xx*x+ }
Early Dial : No
Refer-To Use Target Contact : No
Auto Answer : No
Send Anonymous : No
Anonymous Call Rejection : No
Call Log : Log All
Special Feature : Standard
Feature Key Synchronization : Disable
Enable Call Features : Yes
Call Forward Unconditional :
Call Forward When Busy :
Call Forward When No Answer :
Delayed Call Forward Wait Time (s) : 20
No Key Entry Timeout (s) : 4
Ring Timeout (s) : 60
Transfer on Conference Hangup : No
Use # as Dial Key : Yes
Account ring tone : Ring tone 1
Advanced Setting - General Settings
Local RTP Port : 5004
Use Random Port : Yes
Disable PC Port : No
Disabling in-call DTMF display : No
Keep-alive Interval (s) : Yes
STUN Server : stun.ipvideotalk.com
Use NAT IP :
SIP TLS Certificate :
SIP TLS Private Key :
SIP TLS Private Key Password : (Wachtwoord)
Call Features
Disable Call-Waiting : No
Disable Call-Waiting Tone : No
Disable DND Reminder Ring : No
Disable Direct IP Call : No
Escape '#' as %23 in SIP URI : Yes
Offhook Auto Dial :
Ring Tone
Dial Tone : f1=350@-13,f2=440@-13,c=0/0;
Ring Back Tone : f1=440@-19,f2=480@-19,c=2000/4000;
Busy Tone : f1=480@-24,f2=620@-24,c=500/500;
Reorder Tone : f1=480@-24,f2=620@-24,c=250/250;
Confirmation Tone : f1=350@-11,f2=440@-11,c=100/100-100/100-100/100;
Call-Waiting Tone : f1=440@-13,c=300/10000-300/10000-0/0;
PSTN disconnect tone : f1=480@-32,f2=620@-32,c=500/500;
Default Ring Cadence : c=2000/4000;
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