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I also have AudioCodes SIP ATAs in my mixed Cisco/Avaya CS1000 environment. All of the AudioCodes boxes are configured as SIP endpoints in the Aura Session Manager. If I want to configure a DID out of one of the Cisco sites, I just create a translation pattern in CM to send the call over the...

I believe the 3904 had an option where you could specify the local area code, and the phone would use that information to match incoming local calls that included the area code. On the 1140 (depending on your version of firmware), go to Services -> Telephone Options -> Call Log Options -> Area...

If you are seeing junk characters after connnecting, then the problem may be your terminal settings. Try using 7 bit, Mark Parity, 1 stop bit in your terminal profile and see if that helps.

I haven't dealt with NRS in a couple of years, but you may need to look at the NRS to make sure that there is a valid endpoint associated with dial patterns that begin with '9'.

You can also check your Adaptation Module settings in Session Manager to make sure your Module Parameters are correct. Not sure if this will help, but it doesn't hurt to look (in my enterprise, all of the CS1000 Adaptation Modules have 'fromto=true' in the Module parameter field).

You should go to Configuration -> GW and IP to IP -> Manipulations -> Dest Number Tel->IP and enter your dial patterns that begin with '9'. Here is the code block from my board.ini file (Lines 6 & 7 are for CDP and UDP dialing for my network): [ NumberMapTel2Ip ] FORMAT NumberMapTel2Ip_Index =...

I would check your DSP channels to make sure all are acquired and working properly.

On my systems, the error you are getting is only output when something is dialed that cant be routed. Do you have any other SPNs defined in LD 90 that begin with 2?

You need to create a DMI that sets the CTYP to INTL. Assign this DMI to your RLI and your calls should be markes as international.

I think what that is telling you is that the digits that caused the overflow tone is: 92700#

Assuming you have a signaling server running SIP Line Gateway, then you could probably use NAT to translate an external IP to the internal IP of the SIP Line Gateway node. Just be aware that doing this could expose your signaling server to security risks.

The TGAR on your test phone is 01 and the TARG on your Route is also 01. This is probably why your outbound test call is never hitting the route. You should either remove TARG 01 from the RDB or change the TGAR of the test phone.

Sorry, Voice Item Maintenance probably won't work for you in this case. But you could allow this group some limited administration access to CallPilot, then allow them to modify the SDN table for their main number.

I would setup 2 applications in the CallPilot: - Menu application would be used to send calls to Menu. - Answering-Service application would be used to send calls to Answering Service. Configure Voice Item Maintenance in Callpilot to allow the users to select which application is associated...

Would you mind posting the RDB for your route 10? It may not hurt to also check the Media Security settings in the CS1000 to make sure that one side is not expecting some type of encryption.

Change the Class of service when the room is checked out to a setting that will allow calls to internal extensions.

Are these internal calls or external calls? If a trunk is involved, is something happening to the DSP channels that terminate the trunk? Any additional detail you can provide will help people here make an intelligent diagnosis.

LD 117 RST DN - Reset IP sets with specified DN RST TN - Reset IP set with specified TN RST ZONE - Reset Ethersets in specified zone

I agree with bignose21 above. I would suspect some type of security scan, as those seem to be all the rage these days, especially since it is happening on a somewhat repeatable schedule.

You can set the length of the VNR expected digit lengths using the following subprompts under the VNR prompt: FLEN 1-(16): Flexible length of digits expected CDPL 1-(10): Flexible length of VNR CDP UDPL 1-(19): Flexible length of VNE UDP I would try changing the FLEN and CDPL values first.