Instructions to connect Grandstream 2020 and Ondo?

oziworld · Post by **oziworld** » Thu Feb 26, 2009 11:49 pm

1. Brekeke Product Name and version:

2. Java version:
1.4

3. OS type and the version:
windows
4. UA (phone), gateway or other hardware/software involved:
1.5
5. Select your network pattern from http://www.brekeke-sip.com/bbs/network/ ... terns.html :

6. Your problem:
What are the instructions to connect a grandstream 2020 and Ondo bss?

Harold · Post by **Harold** » Fri Feb 27, 2009 1:22 am

oziworld,
Your multiple posts seem SPAM...

Did you set the SIP Server's IP address as a proxy server at grandstream's settings?
If your SIP Server uses the INVITE authentication, you need to set correct username/password also at the phone.

oziworld · Post by **oziworld** » Fri Feb 27, 2009 3:27 am

These are the settings we gave our client although i am not sure it is correct they can make calls without any problems althoght they dont seem to be able to recieve calls all the time. They can sometimes.

This is the VOIP settings we told them to use:

Web » Advanced Settings »
• Silence Suppression: No
• Voice Frames per TX: Default
• Layer 3 QoS: Default
• Layer 2 QoS: Default
• No Key Entry Timeout: 4
• Use # as Dial Key: No
• Local RTP Port: 5004
• Use random port: No
• Keep Alive Interval: 20
• Use NAT IP: Leave Blank
• STUN server: Leave Blank
• Firmware Upgrade: Default
• DTMF Payload Type: 101
• Syslog Serevr: Leave Blank
• Syslog Level: Default
• NTP Server: Default
• Distinctive Ring Tone: Default
• Disable Call Waiting: Default
• Lock Keypad Update: Default

Web » Account 1 »
• Account Active: Yes
• Account Name: M'ship Number
• SIP Server: voip.delacon.com.au
• Outbound Proxy: voip.delacon.com.au
• SIP User ID: M'ship Number
• Authenticate ID: M'ship Number
• Authenticate Password: Password
• Name: M'ship Number
• Use DNS SRV: Default
• User ID is Phone Number: Default
• SIP Registration: Yes
• Unregister On Reboot: Yes
• Register Expiration: 3600
• Local SIP Port: 5060
• NAT Tranversal: No
• Subscribe for MWI: Default
• Proxy-Require: Leave Blank
• Voice Mail UserID: Leave Blank
• Send DTMF: via RTP (RFC2833)
• Early Dial: No
• Dial Plan Prefix: Default
• Enable Call Features: No
• Session Expiration: 1800
• Min-SE: 900
• Caller Request Timer: Default
• Callee Request Timer: Default
• Force Timer: Default
• UAC Specify Refresher: Default
• UAS Specify Refresher: Default
• Force INVITE: Default
• Enable 100rel: Default
• Account Ring Tone: Default
• Send Anonymous: No
• Auto Answer: No
• Preferred Vocoder: G729A/B(1) GSM(2) PCMU(3) PCMA(4)
• Special Feature: Standard

This is what happens when we try to call their SIP phone.

Session ID : 58
from-url sip:299944155@voip.delacon.com.au
from-ip 202.126.105.100:26768
from-if 220.233.132.76:5060
to-url sip:29994400@220.233.132.76:5060
to-ip 150.101.166.59
to-if 220.233.132.76:5060
call-id NWNjMzYxNDA4YmNjMGViM2I2ZTMxNDdhYTAxMGM1NmI.
rule registered=sip:29994400(sip:29994400@192.168.0.102:5060) & Internal Dialing
plugin InviteSession
sip-packet-total 1
listen-port 5060
sip-packet-stacked 0
phase Inviting
time-inviting Fri Feb 27 18:44:52 EST 2009
rtp-relay on
rtp-srcdst
rtp-dstsrc
media audio
payload -
status active
listen-port 10004
send-port
target 202.126.105.100:40230
packet-count 0
packet/sec 0
current size 0
buffer size 260

Any ideas?

Harold · Post by **Harold** » Wed Mar 04, 2009 12:42 am

Is the phone behind a NAT?
Do you get any error response if a call fails?

oziworld · Post by **oziworld** » Wed Mar 04, 2009 5:37 pm

Yes to both questions.

Harold · Post by **Harold** » Wed Mar 04, 2009 10:13 pm

what kind of error code did you get?