[Solved] Voice not heard on LinPhone

Asterisk Asterisk Support
1

Hi

I have installed Asterisk 11.4 on red hat linux.

I am using both LinPhone and XLite.

call initating Phone call receiving phone status
XLite XLite ringing sound is heard, no voice
XLite LinPhone ringing sound is heard, no voice
LinPhone LinPhone rining sound is heard, no voice
LinPhone XLite rining sound and voice are heard

in XLite, I have selecte udp as the transport, domain : XX.XX.XX.XX:5060

In LinPhone, network protocal SIP(UDP) , port 5060 is selected, Audio RTP/UDP : 7078 are selected.

Could you please tell me if I am missing anythign in the configuration?

2

Not unless you provide the complete configuration and relevant network structure.

3

Thansk David. Please find all the details.

sip.conf

[praveena]
type=peer
secret=praveena123
host=dynamic
port=5060
context=mci

[dinesh]
type=peer
secret=dinesh
host=dynamic
port=5060
context=mci

[naresh]
type=peer
secret=naresh
host=dynamic
port=5060
context=mci

[RETAIL]
type=peer
secret=RETAIL
host=dynamic
context=mci

extensions.conf

[mci]
exten => 6001,1,Dial(SIP/praveena,20)
exten => 6002,1,Dial(SIP/dinesh,20)
exten => 6003,1,Dial(SIP/naresh,20)
exten => 6004,1,Dial(SIP/sandip,20)
exten => 18883525,1,Dial(SIP/BILLING,20)
exten => 18883526,1,Dial(SIP/TECHNICAL,20)
exten => 18883527,1,Dial(SIP/GLOBAL,20)
exten => 18883528,1,Dial(SIP/RETAIL,20)

XLite (4.5 version) setup
Account name:RETAIL
UserId: RETAIL
Domain: :5060
Password: RETAIL
Display name: RETAIL
Authorization Name: RETAIL
Selected Domain
Under Transport selected UDP

LinPhone (3.5.2 version)
Network Settings Tab:
Transport
Set Maximum Transmission Limit - not selected
Send DTMFs as SIP Info - Not selected
Use IPV6 instead of IPV4 - Not selected

Network Protocol and portsSIP(UDP) 5060 is selected
Audio RTP/UDP : 7078
Video RTP/UDP: 9078

NAT and Firewall:Direct Connection to the Internet

Manage SIP Accounts Tab
Diaplay name: praveena
username: 6001
proxy accounts : Added praveena account

4

Assuming you are using the sample RTP configuration files and everything is on one subnet, there is nothing obviously missing, so I would look for things that shouldn’t be there, like firewall rules.

Also you should provide “sip set debug on” output, showing the SDP handshakes (but not noise, like OPTIONS).

5

Thans David, actually problem was with one of the head phones…
Now it works both XLite and LinPhone.