v4.0 SIP Trunk Examples

[Drew2400]

Okay, so here's the deal. I consider myself reasonably intelligent. Less so after a few martini's, but that's another story.

So, I've been trying to figure out how to configure SIP trunks (yes, I'm licensed and have a VCM 8). I just can't get it.

The two I'm pretty interested in are DIDWW and FreeWorldDialup. However, I'll configure Vonage (the softphone later).

Could someone provide a set of examples as to how to configure these? I would be eternally greatfull. And now, the aforementioned martini's are calling my name. %-)

 

[technicaluser4]

If you are using DIDWW you do not need SIP trunks. Goto the lines and create a 2 new IP Lines with the default settings.
In the ip address fields put 212.150.36.116 for one line and 204.11.194.34 for the other line.
These are the 2 sources from which the IP Office will receive the calls.

Your IPO is ready to receive calls. Now you have to make sure that there is a network path from your IPO to the outside world. You have to check routing and access on your firewalls. To cut this short, connect your IPO on the public internet for a while and assign a fixed public ip address.

Goto the didww website, purchase/trial a number and then type the followng in the URI field:
h.323 / 213@194.124.123.122

i.e. extnno/your public ip address.

It will work straight away.

Cheers.

 

[Drew2400]

That's exactly what I did do. However, there is a little anomaly that I'm experiencing. Touch tones/DTMF do not work.

So, if I were to get someone's mailbox and wished to 0 out or *7. This is not possible. I've tried changing from in-band to out-of-band DTMF to no avail.

I should've also eluded to the fact that I have a PIX 506E for a firewall and have been playing with the SIP fixup's (both on and off).

In short, my thoughts were that perhaps DTMF would be supported correctly on the SIP side as opposed to the H.323 side.

Take care...

 

[technicaluser4]

I do not have problems with DTMFs, perhaps did you by any chance increase the volume in db? on your ip phones?
Once I had an issue with DTMFS following an increase in the volume.

I would also skip the PIX for a couple of minutes and check if its blocking something.

 

[Drew2400]

Okay... So, FWD is working. Albeit for the outgoing only.

So, I've turned off fixup for SIP in the Cisco PIX (it messes with the headers). I've forwarded ports 5060,5061,5082,8000 to the IP Office (I'm erroring the more ports the better side).

Outgoing, no prob. I click on the test "Call Me" option on FWD. Nada. Am I forgetting to forward a port? The IPO is set for "Port Restricted Cone NAT" which it determined from FWD's STUN server.

Do I need to forward any more ports?
Thanks,
Drew

 

[Drew2400]

Okay, still self inducing trauma upon myself.
Outgoing to FWD- got it!
Incoming - NO GO.

Forward ports, fixup on/off, etc.

Any ideas?

My thoughts with regards to starting this thread was that in Avaya Tech Bulletin 80, it does list FWD as a "tested" ITSP.

So, what would be the sample IPO line configuration? If it's been tested, it must work, right?

Much thanks...

 

[intrigrant]

This may help:

http://www.cisco.com/univercd/cc/td...2/122newft/122limit/122y/122yu11/ft_fwsip.htm

http://www.sipcenter.com/sip.nsf/html/SIP+Firewall+and+NAT+Traversal

 

[Drew2400]

Okay, so I've narrowed it down to a Pix related issue. Has anyone configured a pix for use with the IPO?

If so, an attached config would be absolutely splendid.

Any suggestions?

Thanks
Drew

 

[Drew2400]

FYI for everyone:

Thanks for the help. I have successfully setup the IPO (at least for testing purposes) to DIDWW (using SIP so the touchtones work), CallCentric (outgoing only at this time), an office IPO (403 v3.2) via H.323 (and haven't noticed a problem as of yet) and lastly FreeWorldDialup.

It's working splendidly all the way around. I had some issues with the PIX which I resolved (NO FIXUPS! and forwarding port 5060 ONLY).

I actually have my cell phones voicemail delivered to my IPO via DIDWW (which is pretty damned fast) to a Start Point in VMPro. And yes, I get email delivery of voicemail.

If anyone has successfully utilized other ITSP's, let me know. I can provide a brief description of the settings if anyone needs to get the above listed ITSP's connected.

Take care,
Drew

 

[MrIPO]

I have just posted some examples for you on the website below.
R

http://www.iporesource.com

"Launching late 2006

 

[Drew2400]

Okay, so, I now have IPO v4 working with CallCentric, FWD, and DIDWW. I still have two questions though.

1) The Calling Party Number/CallerID comes across with the full URI. This makes calling the party back difficult. In the case of CallCentric the number they provide starts with 1777, so, this is not a problem. I have an ARS shortcode setup with 1777N; DIDWW, however, comes across with the URI format (i.e. 4135551212@xxx.xxx.xxx.xxx). If I call the party back using Log or the Missed tab in PhoneManager Pro, it attempts dialing the entire string (all the numbers including the IP numbers). FWD, I prefix with a 393 (F(3),W(9),D(3)). I'd like that to be prefixed to the incoming CallerID from FWD to return the call.

2) Let's say I want to actually call a URI by name. What should I setup for an ARS then? I could use CallCentric to place the call.

Also, if I were to NOT use CallCentric, could I set the IPO to resolve a dial directly. And lastly, yea I know 2 questions, I could conceivably put the IPO on the net and allow direct calling, couldn't I? How would I keep toll fraud down to a minimum?

Thanks one and all again.

 

[DavidCT]

I could use some help setting up with Callcentric.

What are the settings? I'm either missing something really obvious, or...

I would like to trial both incoming and outgoing with them.

 

[Drew2400]

Here's what i've done for CallCentric.
1) Setup and account (yes I know this is obvious, however, you'll need to use the 1777 number they give you.
2) Run the Stun test under System>Lan1 and insure that it is one of the acceptable NAT's (adjust firewall as necessary, mine is a PIX 506e).
3) Create a line entry using the following settings:
SIP Line:
ITSP Domain Name: callcentric.com
ITSP IP addr: 204.11.192.23
Pri Auth Name: your 1777 number
Pri Auth Password: you callcentric PW
Pri Reg Expiry: 29 (that's what I use otherwise I get an error after a while)
Reg Required: yes
In Service: yes
Use Tel URI: no
Voip Silence: no
Re-invite: yes
Compression mode: automatic
Net Config:
UDP
Lan1
5060
5060

SIP URI:
Local URI: Use Auth
Contact: Use Auth
Display Name: Use Auth
Reg: Primary
Inc/Out Groups: whatever (for this example I used 5)
Max Channels (if you have more TDM phones then IP, keep the number a fraction of the total number of VCM channels)

ARS:
177N;/Dial/5
9N;/Dial/5

Now the fun part, the incoming.
Incoming Route:

Bearer: Any Voice
Group: 5
Incoming number: 1777xxxxxxx
Destination: wherever you like

and again for each and every incoming DID number (assuming you purchased it):
Group: 5
Incoming number: 16175551212
Destination: wherever you like

Works like a charm! I hope this helps.
Drew

 

[DavidCT]

Hi Drew,

Thanks for the great details. I set up exactly as you listed and I'm still having problems. Call Centric has been helpful to, here's what they are saying:

--------

The major problem I see there is that the URI:
INVITE Tel:+17771234567 SIP/2.0

and the TO:
To: Tel:+17771234567

are in "Tel" format instead of a SIP URI. This format is not supported by Callcentric, or any other ITSP I'm aware of.

They should look like this:
URI:
INVITE sip:17771234567@callcentric.com SIP/2.0
TO:
To: <sip:17771234567@callcentric.com>

Based on the screenshots you sent i would guess the reason that your Avaya is sending the "Tel" format is because within the SIP line configuration you have "USE TEL URI" checked. If you uncheck it I would guess that it will use the normal and much more common SIP URI format.

Any ideas?

 

[Drew2400]

Yes, you are absolutely correct. I tried turning it on after I got everything working to see what would happen (World hates a coward)! And, as you guessed it, it broke.

I also made a little booboo with the shortcodes. The ars should be

9N;/Dial/N"@callcentric.com"

You do, apparently need the @callcentric.com attached or the call will also not go through.

Now, the one last annoying thing is that I'd like to be able to call the part back by using the logs on the phone or call manager. When it dials the phone it tries to dial the entire phone number and ip address.

It makes it difficult to call people back. I'd also like to attach a code onto the beginning of the incoming call so as I can easily call them back.

I use 393 for FWD. I'd like to prefix 393 to the beginning of the FWD caller to call them back like the prefix for analog circuits.

Hmmmm, yet another thing to ponder.

 

[DavidCT]

Drew,

Is your set up working?

I went in to the ARS for:

177N;/Dial

and changed it to:

177N;/Dial/N"@callcentric.com"

I still get the same recording about the number I called not being in service.

After changing the ARS as you suggested, I noticed in call status that the number dialed shows up as 71234567@callcentric.com (the 177 are not being displayed).

 

[DavidCT]

Alright, I made some progress, but when I get the test call to go through to tellme, it's so garbled I can't make anything out.

The ARS code needs to look like this:

177N;/DIAL/177N"@callcentric.com"

Without the 177 before the N, the system strips out the 177 and sends call centric just one 7 followed by the rest of the number.

 

[Drew2400]

Yes, you would be absolutely correct!

A clear and distinct depiction of the constant turmoil that's going on in my head! Even I find it scary.

What kind of firewall are you using? My only thought is the packets are either really out of order or you don't have enough bandwidth. BTW, that tellme thing really doesn't work too well. I have a few questions that might help:
Which codec do you have selected? And what type of phone are you calling from? What type of VCM is in the system?

There really aren't any good ways to test the throughput, short of actually calling someone.

 

[Drew2400]

Has anyone figured out how to strip off the ITSP part of the URI?

Recalling numbers is a bit of a problem. The phone tries to dial the entire string of numbers. Also, Phone Manager Pro just seems to take a nap when you double click on an entry in the log.

Take care all.

 

[DavidCT]

I'm using Sonicwall, and after working with an Avaya Tech on the SIP team I found two some-what obscure VOIP settings on the Sonicwall which had to be tweaked (and were poorly documented by Sonicwall).

Once I tweaked the Sonicwall, I was able to register with Callcentric, though I still cannot make a call to their test # (17771234567) because the audio is all garbled.

I am able to make calls with AGN, but not receive. If/when AGN responds to my open ticket I hopefully will get it to work.