Hi James,
We have done something similar, but for whatever reason had to use two # signs to terminate the dial string in some cases. We were simply going across h.323 trunks from one CM to another CM. It sounds like you're waiting for stutter dial tone and then entering digits to choose a zone, is that true? Is that a function of your amplifier or CM?
For your call-vector approach, have you provided a wait step to provide answer supervision? If so and it's not working, I've had some success in first establishing audio before performing the route-to step. I've used a recording which is about 500ms of silence to do this. Sounds like voodoo, but in one case it allowed me to preserve the callers original number when routing to an external provider. Without it, they always got the CID of my vdn number.
Now that we're a few years down the line from our SIP conversions, we're taking a different approach to paging. We're adopting Algo 8301 devices which are SIP endpoints for loudspeaker paging and treating them like SIP stations.
Anyone else have a take on this?
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Tom Lynn
Forum Moderator
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Original Message:
Sent: 07-27-2023 05:47 PM
From: James Davis
Subject: 3rd Party SIP Trunk needing to dial Loud Speaker Paging TACs (Trunk Access Codes) in Avaya CM
So we have a bit of a weird one here, we are upgrading a system that is used by our Medical Call Center (AmTelco) and we are going SIP trunking with it. We have everything setup and in place, calls can go back and forth between the Avaya SM/CM (8.1.3.4) system and the AmTelco with no problems, Audio is there and it looks like everything is working.
The only issue we have come up with is, they need to be able to do Overhead Paging in CM, which is via a TAC code.
We have those TAC codes routed correctly in SM, and we see them hit CM when dialed by AmTelco, and it appears to get tied onto the Aux-Trunk port, but the caller never gets the stutter dial tone to enter paging zones. It just gives them dead air. The same call from a station on CM results in Stutter dialtone as expected.
We have tried various options - Dialing the TAC Code, inserting a pound at the end of it. Dialing through the dead air to see if it will just go, but nothing seems to give us back the stutter dial tone we are looking for or allow for a page to go out. We have even tried building a VDN that is dial able by the AmTelco system, and have the Vector either route to number we define OR to digits dialed (Which would be the TAC code), and the same thing happens.
The old AmTelco system did this function pretty much the same way, but over a QSIG ISDN Trunk Group, so we are suspecting we are missing something on the SIP Trunk provisioning (Not like there is a thousand options there to check :) )
We are kind of stumped now as to why we see the call where it needs to be, but we can't get the audio responses we are looking for.
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James Davis
Lead Voice Engineer
University of Nebraska Medical Center
Omaha NE
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