Alcazar setup problem and Peer RTP Address
Phillip:
Hi OBi experts,
With the help of ianobi, I have set up Alcazar as a Toll Free Terminal provider (no registration) on sp2 and a rule to intercept 411 calls, substitute 800FREE411 (8003733411) and direct it to sp2. The SIP handshaking seems to be working OK, but the Peer RTP Address makes no sense. It is not the Alcazar IP nor is it my Public IP. The Peer RTP Address and port number changes with every call and the port number is all over the map (pun intended).
Here are a couple of samples of a Call Status screen:
AND
Ianobi is going to be very busy over the next few days, so I have come to see if any of you have encountered this same situation. Obviously, RTP is the problem; the call connects but there is no voice. But how do I go about addressing it? Any suggestions would be appreciated.
Btw, I expect a new router to be delivered in the next few days, so I will be re-configuring my network anyway. The sky is the limit as they say. I'll have plenty of latitude for testing.
Thanks,
Phillip
Phillip:
I added some relevant info to my sig. Thought perhaps a little config info might help here too:
Alcazar setup on sp2:
Service Providers -> ITSP Profile B -> SIP -> ProxyServer : tollfree.alcazarnetworks.com
Service Providers -> ITSP Profile B -> SIP ->ProxyServerPort: 5060
Voice Services -> SP2 Service -> Enable : (checked)
Voice Services -> SP2 Service -> AuthUserName : any letters/numbers
Voice Services -> SP2 Service -> X_RegisterEnable : (unchecked)
Voice Services -> SP2 Service -> X_ServProvProfile : B
Voice Services -> SP2 Service -> CallerIDName : Whatever
Setup Wizard>Outbound Settings>ITSP DigitMap>
(<**2>(1800|1888|1877|1866|1855)[2-9]xxxxxx|<**2>(800|888|877|866|855)[2-9]xxxxxx|1xxxxxxxxxx|<1>[2-9]xxxxxxxxx|<1281>[2-9]xxxxxxS4|011xx.)
Setup Wizard>Outbound Settings>Phone DigitMap>
([1-9]x?*(Mpli)|[1-9]S9|[1-9][0-9]S9|911|411|**0|***|#|**1(Msp1)|**2(Msp2)|**9(Mpp)|(Mpli))
Setup Wizard>Outbound Settings>Phone OutboundCallRoute>
{(<911:7136259911>):sp1},{(<411:18003733411>):sp2},{(<#:>):li},{([1-9]x?*(Mpli)):pp},{**0:aa},{***:aa2},{(<**1:>(Msp1)):sp1},{(<**2:>(Msp2)):sp2},{(<**9:>(Mpp)):pp},{(Mpli):pli}
Setup Wizard>Inbound Settings>Phone InboundCallRoute>
{ph,pp(290111111)}
Service Providers>ITSP Profile A General>DigitMap>
(<**2>(1800|1888|1877|1866|1855)[2-9]xxxxxx|<**2>(800|888|877|866|855)[2-9]xxxxxx|1xxxxxxxxxx|<1>[2-9]xxxxxxxxx|<1281>[2-9]xxxxxxS4|011xx.)
Service Providers>ITSP Profile B General>DigitMap>
((1800|1888|1877|1866|1855)[2-9]xxxxxx|(800|888|877|866|855)[2-9]xxxxxx)
Hope this helps. Me not so much. ;) While I am learning more every day, I have reached the level of proficiency one might describe as dangerous. :)
Thanks in advance,
Phillip
Phillip:
Nothing? Really? ???
infin8loop:
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.com
X_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.
Phillip:
Thanks infin8loop, I'm starting to see the wisdom of your approach, though I probably look like a calf confronted by a new gate when I try it. ;) I keep telling myself that I'll get the hang of this stuff.
My needs are modest right now. But that could change in the near future. I'm just doing the 800 termination because of the GV inability to handle some 800 #s. Otherwise sp2 would still be unassigned. I thought this setup might be easier than setting up voice gateways. ianobi has been a big help in getting me this far. Been more of a learning exercise than anything. I'll try the STUN settings first and report back.
I have also been told on another site that the U-verse (Underverse?) 2WIRE 3600HGV gateway may be blocking access to the OBi RTP communications. I tried to set it up as suggested. But it wouldn't accept the password, so firewall changes are currently blocked. Seems cmos is going bye bye. Even Humpty-Dumpty, I mean AT&T tech support, couldn't get it to behave even when using the back door. There's a new gateway en route.
I am really going in the hole here. Pizza and beer for QBZappy, ianobi and several others. Now I guess I'm bringing the Crown Royal too. Would you like to share a little Makers Mark instead? ;D
Navigation
[0] Message Index
[#] Next page