SIP Trunking with IP Office and Call Centric

[superikey]

Need some help with this.
I followed someons drew2400's instructions on

http://www.tek-tips.com/viewthread.cfm?qid=1341338I

am first trying to geti ncomming working but no go.

When I call the Call Centric # I get "your call can not be completed as dialed...."

I am using a GTA, GB-1200 Firewall.
I also tried opening ports 5060 to the IPO
I also set 5700 as the Public Port for the STUN Settings and opened that port in the Firewall as well.

Any ideas?

Isaac Braca
CTO / ICCS & Co., LLC.
Email: braca at iccsllc.com

ICCS, Your Premier IT Partner, is a NYC Based Avaya Business Partner and IT Consulting Firm.

 

[IPOfficeIreland]

hi issac
1st thing to do is open Monitor to check if the account is registering with the sip provider. Some sip providers web interface will tell you "on line or not".

Incoming calls are the easy part on avaya sip.
1 Valid SIP Trunk License
2 VCM Channel Free
3 Create New SIP Line
4 Enter registration user name and password (if reg required)
5 Add SIP uri - default settings entered are usually fine
6 Give the SIP line a unique incoming and outgoing line id
7 Save and Restart
8 Open Monitor, uncheck all tabs except SIP and Print.
9 Test and good luck

 

[Drew2400]

Well... TheTaker is right. Open monitor and check to see if you are registering. You should see TX (blah blah) REGISTER. Then you should see RX (blah blah) OK. Make sure options in the SIP tab for tracing are enabled.

However, I believe you problem may be in a short code. The fact that you are actually getting the CallCentric voice (that's a callcentric message), means that you should be registered. I would believe that it's your shortcode.

Make sure that you are dialing the FULL 11 digit number (16175551212, not 5551212 or 6175551212). Also, as the bald spot in the front of my head indicates (banging my head against the wall), your shortcode has to have a ;
Example:
SC: N;
Dial: N"@callcentric.com"
DIAL
Line Group: (whatever)

Also make sure Tel URI is NOT checked. I can't even calculate the number of hours I spent trying to figure out what was going wrong. The semicolon indicates that the IPO has to wait until all digits are dialed prior to passing the string onto the destination. BTW, this is not necessary when talking to another IPO.

Another thing I found out about regarding CallCentric was that all incoming numbers are mapped to your 1777 number. So you only need one entry in the SIP URI tab under line config. For instance, I have 2 numbers here, 1 number in FL, an 800# and the 1777 number. All calls are carried over the 1777 number. Make sure you have an incoming call route to distribute the incoming DDI appropriately.

For those interested. I now have CallCentric, FreeWorldDialup, DIDWW, SellVoip (amazingly still around) working the IPO. I wanted a provider that let's me control the outgoing CallerID so that I can utilize twinning and someone suggested CallWithUs. Of which, I can't seem to get working. Let us know if this helps....

Take care,
Drew

 

[DavidCT]

superikey,

Your problem may be related to "SIP Transformations", which basically is translating between the private IP of your IPO, and the Public your IPO appears to be registering from at the VoIP provider.

I have found that if the firewall can do SIP transformations, that's the most reliable way to go. Turn on SIP transformations on the firewall, and then on the IPO under "LAN1", "Network Topology" set "open internet" for the firewall type, use 0.0.0.0 for it's public IP, and set the port to 5060.

If the above does not work, then you can try to open up ports on your firewall. Open port 5060 AND the entire RTP range, 49152 to 53246 from the public IP to the private IP of the IPO. Then run the STUN test on the IPO.