I normally use
sip.tollfreegateway.com but substituted
tollfree.alcazarnetworks.com with equal success:
Terminal ID PHONE1 SP2
State connected connected
Peer Name
Peer Number 18004444444
18004444444@tollfree.alcazarnetworks.com Start Time 00:05:42 00:05:42
Duration 00:00:25 00:00:25
Direction Outbound Outbound
Peer RTP Address 74.120.95.20:17334
Local RTP Address 192.168.0.41:16800
RTP Transport UDP
Audio Codec tx=G711U; rx=G711U
RTP Packetization (ms) tx=20; rx=20
RTP Packet Count tx=1288; rx=1366
RTP Byte Count tx=221536; rx=234952
The Peer RTP Address above is close to the one in your screen grab. I suspect they might be using load balancing across multiple servers.
My SP2 is configured for
voip.ms and the "tollfree" route is configured on Voice Gateway 8:
Enable: checked
Name: Toll Free
AccessNumber: SP2(tollfree.alcazarnetworks.com)
DigitMap: (18(00|88|77|66|55)xxxxxxx|<1>8(00|88|77|66|55)xxxxxxx)
AuthUserID: whatever number entered here is passed as callerid number
AuthPassword: leave default box checked
Add the following to the Phone Port Digit Map DigitMap: |(Mvg8)|
Add the following to Phone Port OutboundCallRoute: ,{(Mvg8):vg8},
I don't have it in other DigitMaps like SP1 and SP2 because then I can use **1 and **2 to route
800 calls through the providers configured on SP1 or SP2 instead if needed. You might consider getting the DigitMap for the 800 number processing out of all the places you have it.
You might consider configuring SP2 as a "dummy" SIP provider by using 127.0.0.1 in the ProxyServer instead of pointing it to alcazarnetworks.
If you still have no voice on RTP stream try configuring a STUN server on SP2:
STUNEnable: checked
STUNServer:
stun.ideasip.comX_STUNServerPort: 3478
If you still have no voice on RTP, then forward (in your router) the RTP ports (probably 16800-16998) on SP2 to your OBi. And disable SIP ALG in the router if it has the option (mine does not).
Updated: Added Phone Port OutboundCallRoute above that I omitted in first post.