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rlg192320
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Telco Intercept messages

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Hello,

I am wondering if there is a way for the Adtran 9xx/e to play telco intercept messages instead of just passing a 503 SIP message down a SIP trunk. I am currently deploying a 908e for a PRI to SIP customer. The customer is receiving a PRI handoff from the Adtran 908e and the 908e has a SIP trunk to an SBC. When the PBX sends a call out, the Adtran receives a 503 for a disconnected number and the customers PBX apparently cannot play an "disconnected" message and instead routes the caller to their own AA.

Thanks,

Robert Garza

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jayh
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Honored Contributor

Re: Telco Intercept messages

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There isn't a recording/playback function in the TA900 series unit.  It sounds as if the PBX may have its own default route to the AA if it receives a rejection from the trunk.  You may need to tweak this behavior in the PBX.  Perhaps a stop-route or not have the AA as default on that trunk.

If you unplug the PRI entirely, does a caller get the AA when attempting to place an outbound call?  If so you'll likely need to find the stop-route command in the PBX or yank the default on outbound calls on that trunk.  Defaulting to the AA is probably desired behavior on incoming calls.

Another possibility:  There is a default mapping between SIP messages and ISDN cause codes.  By default, SIP 503 maps to ISDN 41.

41 - temporary failure [Q.850]

This cause indicates that the network is not functioning correctly and that the condition is no likely to last a long period of time; e.g., the user may wish to try another call attempt almost immediately.

You may try a different mapping to something that will cause the PBX to send its own internal recording or fast-busy instead of punting to the AA.  The following may work to cause the PBX to play a fast-busy or internal recording:


Cause No. 1 - Unallocated (unassigned) number [Q.850]

This cause indicates that the called party cannot be reached recluses although the called party number is in a valid format. It is not currently allocated (assigned).

Cause No. 4 - send special information tone [Q.850]

This cause indicates that the called party cannot be reached for reasons that are of a long term nature and that the special information tone should be returned to the calling party.

Cause No. 17 - user busy [Q.850]
This cause is used to indicate that the called party is unable to accept another call because the user busy condition has been encountered. This cause value may be generated by the called user or by the network. In the case of user determined user busy it is noted that the user equipment is compatible with the call. Note that this is technically incorrect and will cause the PBX to play a normal (slow) busy signal on failed calls but it may be your only choice. Callers won't be able to distinguish between a busy number and a failed call.


To set the mapping you would enter one of the following commands, as appropriate:

voice cause-code-map to-pri 503 1

voice cause-code-map to-pri 503 4

voice cause-code-map to-pri 503 17


Full list of ISDN cause codes here:

http://cnes.com/causecodes.html

I don't see Adtran's default mapping documented anywhere easily searched, but the following command will display them on a working unit.

show run verbose | include cause-code-map to-pri


This is also configurable in the other direction should the need arise to modify the SIP message based on what the PBX returns.  The syntax is:


voice cause-code-map from-pri [ISDN Cause] [SIP message]


Tweak these values with care, the defaults are usually just fine. 

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jayh
Honored Contributor
Honored Contributor

Re: Telco Intercept messages

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There isn't a recording/playback function in the TA900 series unit.  It sounds as if the PBX may have its own default route to the AA if it receives a rejection from the trunk.  You may need to tweak this behavior in the PBX.  Perhaps a stop-route or not have the AA as default on that trunk.

If you unplug the PRI entirely, does a caller get the AA when attempting to place an outbound call?  If so you'll likely need to find the stop-route command in the PBX or yank the default on outbound calls on that trunk.  Defaulting to the AA is probably desired behavior on incoming calls.

Another possibility:  There is a default mapping between SIP messages and ISDN cause codes.  By default, SIP 503 maps to ISDN 41.

41 - temporary failure [Q.850]

This cause indicates that the network is not functioning correctly and that the condition is no likely to last a long period of time; e.g., the user may wish to try another call attempt almost immediately.

You may try a different mapping to something that will cause the PBX to send its own internal recording or fast-busy instead of punting to the AA.  The following may work to cause the PBX to play a fast-busy or internal recording:


Cause No. 1 - Unallocated (unassigned) number [Q.850]

This cause indicates that the called party cannot be reached recluses although the called party number is in a valid format. It is not currently allocated (assigned).

Cause No. 4 - send special information tone [Q.850]

This cause indicates that the called party cannot be reached for reasons that are of a long term nature and that the special information tone should be returned to the calling party.

Cause No. 17 - user busy [Q.850]
This cause is used to indicate that the called party is unable to accept another call because the user busy condition has been encountered. This cause value may be generated by the called user or by the network. In the case of user determined user busy it is noted that the user equipment is compatible with the call. Note that this is technically incorrect and will cause the PBX to play a normal (slow) busy signal on failed calls but it may be your only choice. Callers won't be able to distinguish between a busy number and a failed call.


To set the mapping you would enter one of the following commands, as appropriate:

voice cause-code-map to-pri 503 1

voice cause-code-map to-pri 503 4

voice cause-code-map to-pri 503 17


Full list of ISDN cause codes here:

http://cnes.com/causecodes.html

I don't see Adtran's default mapping documented anywhere easily searched, but the following command will display them on a working unit.

show run verbose | include cause-code-map to-pri


This is also configurable in the other direction should the need arise to modify the SIP message based on what the PBX returns.  The syntax is:


voice cause-code-map from-pri [ISDN Cause] [SIP message]


Tweak these values with care, the defaults are usually just fine. 

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Anonymous
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Re: Telco Intercept messages

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Hello,

I went ahead and flagged the "Correct Answer" on this post to make it more visible and help other members of the community find solutions more easily.  If you don't feel like the answer I marked was correct, feel free to come back to this post and unmark it and select another in its place with the

applicable buttons.  If you still need assistance, we would be more than happy to continue working with you on this - just let us know in a reply.

Regards,

Geoff