Grandstream HT701 configuration
... / Grandstream HT701 configu...
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Question

Grandstream HT701 configuration

by
tuks
Created on 2018-04-07 17:11:06 (edited on 2024-09-04 12:49:42) in Forfaits VoIP

Bonjour,

Après plusieurs tests voici la configuration pour le HT701 (et 702?) qui fonctionne à merveille :

**Dans ADVANCED SETTINGS :**
Keep-alive Interval: 15

**FXS PORT1 :**
Account Active: Yes
Primary SIP Server: sip.ovh.fr
Failover SIP Server:
Prefer Primary SIP Server: No
Outbound Proxy: sip.ovh.fr
Allow DHCP Option 120( override SIP server ): No
SIP Transport: UDP
NAT Traversal: Keep-Alive
SIP User ID: 0033wwxxyyzz
Authenticate ID: 0033wwxxyyzz
Authenticate Password: xxxx
Name: tux
DNS Mode: A Record
Tel URI: Disable
SIP Registration: Yes
Unregister On Reboot: No
Outgoing Call without Registration: No
Register Expiration: 25
Reregister before Expiration: 0
SIP Registration Failure Retry Wait Time: 120
SIP Registration Failure Retry Wait Time upon 403 Forbidden: 1200
Enable SIP OPTIONS Keep Alive: Yes
SIP OPTIONS Keep Alive Interval: 15
SIP OPTIONS Keep Alive Max Lost: 5
Layer 3 QoS: 24-46
Local SIP port: 5060
Local RTP port: 5004
Use Random SIP Port: No
Use Random RTP Port: No
Hold Target Before Refer: Yes
Refer-To Use Target Contact: No
Transfer on Conference Hangup: No
Disable Bellcore Style 3-Way Conference: No
Remove OBP from Route Header: No
Support SIP Instance ID: No
Validate Incoming SIP Message: No
Check SIP User ID for incoming INVITE: No
Authenticate incoming INVITE: No
Authenticate server certificate domain: No
Authenticate server certificate chain: No
Trusted CA certificates:
Allow Incoming SIP Messages from SIP Proxy Only: No
Use Privacy Header: Default
Use P-Preferred-Identity Header: Default
SIP REGISTER Contact Header Uses: LAN Address
SIP T1 Timeout: 0.5
SIP T2 Interval: 4
SIP Timer D: 0
DTMF Payload Type: 101
Preferred DTMF method:
Priority 1: RF2833
Priority 2: SIP INFO
Priority 3: In Audio
Disable DTMF Negotiation: No
Generate Continuous RFC2833 Events: No
Send Hook Flash Event: No
Flash Digit Control: No
Enable Call Features: Yes
Offhook Auto-Dial:
Offhook Auto-Dial Delay: 0
Proxy-Require:
Use NAT IP:
Use SIP User-Agent Header:
Distinctive Ring Tone: Ring tone 1,2 &3
Disable Call-Waiting: No
Disable Call-Waiting Caller ID: No
Disable Call-Waiting Tone: No
Disable Connected Line ID: No
Disable Receiver Offhook Tone: No
Disable Reminder Ring for On-Hold Call: No
Disable Visual MWI: No
Do Not Escape '#' as %23 in SIP URI: No
Disable Multiple m line in SDP: No
Ring Timeout: 60
Delayed Call Forward Wait Time: 20
No Key Entry Timeout: 4
Early Dial: No
Dial Plan Prefix:
Use # as Dial Key: Yes
Dial Plan: { xxx+ }
SUBSCRIBE for MWI: No
Send Anonymous: No
Anonymous Call Rejection: No
Special Feature: Standard
Session Expiration: 180
Min-SE: 90
Caller Request Timer: No
Callee Request Timer: No
Force Timer: No
UAC Specify Refresher: Omit (Recommended)
UAS Specify Refresher: UAC
Force INVITE: No
Enable 100rel: No
Add Auth Header On Initial REGISTER: No
Conference URI:
Use First Matching Vocoder in 200OK SDP: No
Preferred Vocoder:
choice 1: PCMU
choice 2: PCMA
choice 3: G723
choice 4: G729
choice 5: G726-32
choice 6: iBLC
Voice Frames per TX: 2
G723 Rate: 6.3kbps encoding rate
iLBC Frame Size: 20ms
iLBC Payload Type: 97
VAD: No
Symmetric RTP: No
Fax Mode: T.38
Fax Tone Detection Mode: Caller
Re-INVITE After Fax Tone Detected: Enabled
Jitter Buffer Type: Adaptive
Jitter Buffer Length: Medium
SRTP Mode: Disabled
Crypto Life Time: Enabled

SLIC Setting: EUROPEAN CTR21
Caller ID Scheme: Bellcore/Telecordia
DTMF Caller ID: Start Tone Default Stop Tone Default
Polarity Reversal: No
Loop Current Disconnect: No
Loop Current Disconnect Duration: 200
Enable Pulse Dialing: No
Enable Hook Flash: Yes
Hook Flash Timing: 300- maximum: 1100
On Hook Timing: 400
Gain: TX OdB RX -6dB
Disable Line Echo Canceller (LEC): No
Disable Network Echo Suppressor: No
Outgoing Call Duration Limit: 0
Ring Frequency: 25
Enable High Ring Power: No

Ring Tone 1: c=2000/4000;
Ring Tone 2: c=2000/4000;
Ring Tone 3: c=2000/4000;
Ring Tone 4: c=2000/4000;
Ring Tone 5: c=2000/4000;
Ring Tone 6: c=2000/4000;
Ring Tone 7: c=2000/4000;
Ring Tone 8: c=2000/4000;
Ring Tone 9: c=2000/4000;
Ring Tone 10: c=2000/4000;


1 Reply ( Latest reply on 2018-07-18 22:13:31 by
FredericD17
)

Bonjour!

J'ai un Grandstream GXP1100 que j'utilise ici, en Equateur.

IL fonctionne. Ensuite, au bout de quelques dizaines de secondes, il sonne occupé quand j'appelle n'importe quel numéro.

En fait, même quand je retire le câble, il sonne toujours occupé comme s'il était encore connecté! Je fais le numéro et ça sonne occupé :-)

Je débranche, je rebranche et ça fonctionne.

J'ai tout essayé et je reste sans réponse... SI vous avez une idée?

Je pense que c'est une question de session STUN ou UDP mais pas moyen de la changer dans le routeur.

C'est con, avec ZOIPER, tout fonctionne! Sauf que la qualité n'est pas aussi bonne...

Merci!