Call routing from PRI to SIP Trunk

I am attempting to get calls coming in to my Asterisk 1.6 / FreePBX 2.6 server on its Digium TE220B PRI card re-routed back out of the server across a SIP trunk.

Currently, any call originating on the Asterisk box is calling out across the SIP trunk just fine. Any call coming into the box on the PRI is receiving ‘ss-noservice’.

Digium helped me set up the TE220B, so I don’t believe there is a configuration problem there.


channel => 1-23,25-47


loadzone = us
defaultzone = us

Yes, in this scenario, the Asterisk box is supposed to provide the clock across both spans.

In my trunk setup from within FreePBX I’m trying to use groups so I don’t have to configure 46 individual trunks. As seen above, I have group=1. Therefore, I would think that I could create one Zap/DAHDI trunk with Zap Identifier (trunk name) set to ‘g1’ correct?

Although, I’m not even sure that this matters on inbound calls.

Any help would be greatly appreciated.

First, use the Asterisk CLI to see what is coming in on the PRI for inbound number. Make sure it matches your dial plan. Try setting up an extension on the Asterisk and route one of the inbound PRI extensions to it to see if it works.

The SS-No Service is an indication that the Asterisk cannot process the inbound call and sends a reject back to the originator. Check your contexts and dial plan to be sure.

On the trunk, the “context=from-pstn” follows the inbound routes/rules and treats all incoming calls this way. It’s purpose is to route incoming calls to extensions and IVRs.

The “context=from-internal” follows the outbound routes to the SIP trunk. Internal extensions matching the outbound route are sent to the SIP trunk.

If you change the context of your PRI trunk to “context=from-internal”, then incoming calls from the PRI will follow the outbound route and be sent out the SIP trunk.

Moved to a new topic…Sorry

mayankraj -

This is a completely different issue don’t hijack another users thread

“mux converting to IP” means nothing, I have no clue what you are talking about.

The SIP trunk you are trying to use is not setup correctly so Asterisk is returning congested.

Useful information is a SIP debug trace (with verbose turned off), your trunk configurations and a cogent explanation of what you are trying to do.

Start a new thread please.

A short number combination called Feature Codes are brought into play in routing calls through any of the call management features of eConsole. When feature codes are used, the most important advantage that a customer can benefit from is the ability to reconfigure many different numbers simultaneously. There is no need to individually configure numbers which would take so much of your time. The necessary configuration simply needs to be applied to a Feature Code and the settings would be effective after routing your numbers through to the Feature Code.
Furthermore, the use of Feature Codes can be combined with the feature Auto Attendant. Doing so would allow a customer to dial through to IVR Menu, Hunt Group and many other call management features of 0800 and 0845 numbers.
A good example is a large office environment in which dedicated account managers are assigned to attend to customers. Staff members no longer have to give out direct contact numbers but just provide customers with their Feature Code instead. The Feature Code works together with an IVR and Auto Attendant. With the Feature Codes configured, calls received during office hours will be routed directly to them. When calls are received out of office hours, they will be routed through to their personal Voice Mailbox.