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Alcazar setup problem and Peer RTP Address

Started by Phillip, February 19, 2013, 12:56:56 AM

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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
Obi100, sp1 Anveo, sp2 Alcazar Networks Toll Free Terminal provider, Cisco Gigabit modem, TP-Link router/switch

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
Obi100, sp1 Anveo, sp2 Alcazar Networks Toll Free Terminal provider, Cisco Gigabit modem, TP-Link router/switch

Phillip

Obi100, sp1 Anveo, sp2 Alcazar Networks Toll Free Terminal provider, Cisco Gigabit modem, TP-Link router/switch

infin8loop

#3
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.
"This has not only been fun, it's been a major expense." - Gallagher

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
Obi100, sp1 Anveo, sp2 Alcazar Networks Toll Free Terminal provider, Cisco Gigabit modem, TP-Link router/switch

infin8loop

#5
I left out the corresponding entry in Phone Port OutboundCallRoute: ,{(Mvg8):vg8},

Hopefully that's all I left out.

Updated original post as well. I seem to be challenged in getting all the settings out of the Obi and pasted here. LOL


"This has not only been fun, it's been a major expense." - Gallagher

Phillip

I made no other changes but to add the STUN server info. It worked!

So the Maker's Mark is a no go? OK, Crown Royal it is! Thanks.  :)
Obi100, sp1 Anveo, sp2 Alcazar Networks Toll Free Terminal provider, Cisco Gigabit modem, TP-Link router/switch

ianobi

While working on this project, I tried to have smaller/neater digit maps by using a rule such as 1?800xxxxxxx (1 or no 1 followed by 800 etc). This works fine in a digit map on its own. However, when sharing a digit map with rule 1xxxxxxxxxx even if you dial 18002345678 the Digit Map Processor will use rule 1xxxxxxxxxx as the best match. It seems that "1" always takes precedence over "1?". Just goes to show I'm still learning!

Using digit maps inside other digit maps can make them smaller and easier to understand. Taking Phillip's example above:

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)

The first two rules in Profile A DigitMap (Msp1) completely relate to the Profile B DigitMap (Msp2). So we can rewrite Profile A DigitMap (Msp1) as:

Service Providers>ITSP Profile A General>DigitMap>
(<**2>(Msp2)|1xxxxxxxxxx|<1>[2-9]xxxxxxxxx|<1281>[2-9]xxxxxxS4|011xx.)


Felix

Quote from: ianobi on February 20, 2013, 10:03:09 AM
(<**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.)
Alcazar allows toll-free numbers without first 1? I am using Call With Us for toll free dialing (I use them for international dialing anyway; and toll-free is free, so why not). But without 1-8xx I may end up in Cambodia for 855 or in Guangdong, China (for 866). I guess, it's better than Somali where somebody else got when skipping 1... :)

ianobi

Yup, Alcazar accepts 8xxxxxxxxx, 18xxxxxxxxx, +18xxxxxxxxx.

That does mean a slight risk of heading to expensive African destinations! Users could cut out the 8xxxxxxxxx option in their digit maps to be extra safe.

Phillip

Thanks for the heads up on the 1 800 problem, Felix. I do appreciate it. Since my cell service doesn't require a 1, I have gotten into that habit. You may have just saved me a chunk of change! Thanks!
Obi100, sp1 Anveo, sp2 Alcazar Networks Toll Free Terminal provider, Cisco Gigabit modem, TP-Link router/switch

Phillip

I don't think I have this correct.  ???

The following SHOULD require a '1' before any 8xx number and will direct the call to sp2, but the subsequent rules will allow 10 digit dialing and international calls out sp1, right? But what happens when a 10 digit 8xx number is dialed? I would want to prepend a '1' in that case and sent it to sp2, but I'm don't know if this gets it done. HELP!

Service Providers>ITSP Profile A General>DigitMap> <**2>(Msp2)|1xxxxxxxxxx|<1>[2-9]xxxxxxxxx|<1281>[2-9]xxxxxxS4|011xx.)


Service Providers>ITSP Profile B General>DigitMap> (18(00|88|77|66|55)[2-9]xxxxxx)
Obi100, sp1 Anveo, sp2 Alcazar Networks Toll Free Terminal provider, Cisco Gigabit modem, TP-Link router/switch

ianobi

This should work:

Service Providers>ITSP Profile B General>DigitMap>
(18(00|88|77|66|55)xxxxxxx|<1>8(00|88|77|66|55)xxxxxxx)

If you dial 800..., 888... etc the sp2 digit map should match it and prepend the "1". The rule in sp1 digit map <1>[2-9]xxxxxxxxx is a worse match and therefore should not be chosen by the Digit Map Processor.

Try it and check using Status > Call History to make sure it is working as you want.

Phillip

First I made an 8xx (800free411) call sans the '1' and with the old configuration (See post #2):

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)

<CallHistory date="2/26/2013" time="07:39:20">
<Terminal id="PHONE1" dir="Outbound">
<Peer name="" number="**28003733411"/>
<Event time="07:39:20">New Call</Event>
<Event time="07:39:34">End Call</Event>
</Terminal>
<Terminal id="SP2" dir="Outbound">
<Peer name="" number="8003733411"/>
<Event time="07:39:22">Call Connected</Event>
</Terminal>
</CallHistory>

As you can see, it went to sp2 and was connected. Next I made the following changes:
Service Providers>ITSP Profile A General>DigitMap> (<**2>(Msp2)|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)

<CallHistory date="2/26/2013" time="08:48:31">
<Terminal id="PHONE1" dir="Outbound">
<Peer name="" number="18003733411"/>
<Event time="08:48:31">New Call</Event>
<Event time="08:48:40">End Call</Event>
</Terminal>
<Terminal id="GoogleVoice1" dir="Outbound">
<Peer name="" number="18003733411"/>
<Event time="08:48:33">Call Connected</Event>
</Terminal>
</CallHistory>

8xx numbers without a '1' are routed to sp2 and fail with a busy signal. The call is not recorded in the call log.  But the 8xx calls made with a '1' now go out through sp1 (my GV account).

OK found one problem and fixed it! No open parenthesis in ITSP Profile A General>DigitMap. I guess I didn't select the entire line when I did the cut and paste. So I did another round of call with and without the '1'.

WITH a '1' I get the desired result.
<CallHistory date="2/26/2013" time="09:16:35">
<Terminal id="PHONE1" dir="Outbound">
<Peer name="" number="**218003733411"/>
<Event time="09:16:35">New Call</Event>
<Event time="09:16:44">End Call</Event>
</Terminal>
<Terminal id="SP2" dir="Outbound">
<Peer name="" number="18003733411"/>
<Event time="09:16:36">Call Connected</Event>
</Terminal>
</CallHistory>

WITHOUT a '1' the call is routed to Google Voice on sp1. Then I get a recorded message repeating the number dialed  and a notice that the provider rejected the call; reason: error 404.
<CallHistory date="2/26/2013" time="09:16:55">
<Terminal id="PHONE1" dir="Outbound">
<Peer name="" number="18003733411"/>
<Event time="09:16:55">New Call</Event>
</Terminal>
<Terminal id="GoogleVoice1" dir="Outbound">
<Peer name="" number="**218003733411"/>
<Event time="09:16:57">End Call</Event>
</Terminal>
</CallHistory>

So what is actually happening here? I am lost and I really need some help.

What I would like to do is have all 8xx calls go throught sp2. I want all 8xx without a '1' to prepend a '1' to the 8xx numbers so that the call goes through to Alcazar for Toll Free termination and not GV. Am I getting any closer to that goal? I can't tell.
Obi100, sp1 Anveo, sp2 Alcazar Networks Toll Free Terminal provider, Cisco Gigabit modem, TP-Link router/switch

Phillip

Oh! Hi Ian,  :)

I'll try your solution, make a couple of calls and post the results.

Thanks!
Obi100, sp1 Anveo, sp2 Alcazar Networks Toll Free Terminal provider, Cisco Gigabit modem, TP-Link router/switch

Phillip

BINGO! Works like a champ! Thank you sir!  ;D
Obi100, sp1 Anveo, sp2 Alcazar Networks Toll Free Terminal provider, Cisco Gigabit modem, TP-Link router/switch