NEW To SIP TRUNKING Configuration

[erniepangie]

I need help from scratch to configure SIP TRUNKING. We have already purchased SIP Licenses and network provider provided the IP addresses. We have 10 phone numbers dedicated to MAIN number for outgoing and incoming calls.

 

[Okkie26]

1: How do you connect the IPO to the SIP network?

DO NOT connect the IPO directly to the internet.

 

[erniepangie]

THE IPO is connected to the LAN PORT. I can configure remotely the Avaya IP Office through VPN connectivity.

 

[Okkie26]

But how are you going to connect to the SIP provider?

Do they offer a dedicted SIP router? Or is it an open intetnet connection?

Are you able to ping the SIP providers IP address from the IP Office using system status?

Do you have any credentials given from the SIP provider?

Do you have any basic knowledge of SIP?

 

[erniepangie]

They installed a dedicated SIP router in our data center room. Yes, they have provided us credentials. I don't have much knowledge about SIP. But I understand how it works. But the configuring the Avaya IP Office is too complicated for me. I can PING the SIP Provider IP address from command prompt but not from SYSTEM STATUS.

 

[tlpeter]

Probably because there isn't an IProute configured for this.
But SIP isn't easy and never the same so this will be very hard to get working if you do not have enough knowledge with programming an IPOffice.


BAZINGA!

I'm not insane, my mother had me tested!

 

[Okkie26]

So, setup an IP route first.

Then try to ping to the SIP provider.

 

[erniepangie]

Do I need to do this in the command prompt executing ROUTE ADD or from the Avaya IP Office Manager?

Here is the info provided by the network or ISP.

I will change the IP addresses into something else for security purposes:
================================================================================
Kindly consider the following while configuring your PBX:

*For signaling: use IP : xx.xx.xxx.70 ( SIP ) on PORT 5060

*Regarding Numbering Format, use the following:
• For outgoing Calls :
The originating Number (A#), should be 96614510XXXX format.
The Destination Number should be 0NXXXXXX (N area code) or 00XXXXXXXXX (for international)
• For incoming Calls:
The Destination Number (B#), should be 014510XXXX Format.
The originating Number (A#), will be 0NXXXXXXX or 00XXXXXXXXXXX Format

*Use Audio Codec's G711-aLaw & G729
*Use T.38 For FAX
*set DTMF to RFC2833
*Make sure to reply with 200Ok for our OPTIONS messages ( ping messages for the SIP)
*configure the following SIP Timers: “Min-SE=1800 “’ & “Expires=300”


Your Numbers are:

GTS: 966xxxxxxxx0 - 966xxxxxxxx9; (10 Numbers)

For connectivity consider the following:

SIP CE: xx.xx.x.42
SIP GW: xx.xx.x.41
Subnet mask: /30
SIP VLAN: 1225

Notes:
Kindly make sure to have GO SIP GW (xx.xx.xxx.70) routed to (xx.xx.x.41) as the next-hop.
Kindly make sure to have SIP CE IP addresses are in VLAN 1225

 

[Okkie26]

IP Office manager.

The IP Office need to make contact to the SIP provider.

 

[derfloh]

I've hardly seen such detailed information from a SIP provider...

 

[amriddle01]

The system doesn't do VLANs however

 

[Okkie26]

No, correct.

If needed, put a switch between de router and the IPO, and set the switchport for the IPO to untagged vlan 1225

 

[derfloh]

I would think the provider will deliver a router that will have an untagged interface internal and will tag the voice packets on it's WAN side.

 

[Okkie26]

Thats why I said.. If needed..