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Old 01-08-2009, 12:43 PM   #1
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DTMF Site to Site issue

I have a customer that has a conference bridge 7.1 at one location. We are trying to have a dial in number from both of their locations, but when we dial the bridge from the remote site, the bridge will not accept the DTMF tones. When i monitor on the Trunk Test Tool it keeps showing:
09:28:24.132: Dial "17028734100" Flags: 80
09:28:47.930: Dial "17028734100" Flags: 80

The basic setup is a WG that has the conference ports at the main site assigned with a local DID that works great. I then havce tried route points, additional WG's, and even a phantom extension to call into the WG from the remote site, using on of the remote sites local DID's. Nothing seems to be working. Any ideas?
Thanks in advance!!
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Old 01-08-2009, 12:56 PM   #2
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Update: Internal calls will not transfer DTMF across the network either at this point, at least to the Conference Bridge
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Old 01-08-2009, 02:45 PM   #3
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which codec are you using intrasite and intersite? if it works intrasite maybe try changing the intersite to 711 and see if that works. could be an inband vs out of band dtmf problem.

my suggestion is not so much as a long term fix but as a test.

matt
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Old 01-08-2009, 05:24 PM   #4
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On the conference bridge, look under the "Configuration" tab
then "VOIP Settings" What do you have in these boxes?
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Old 01-13-2009, 03:27 PM   #5
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We have tried everything from low bandwidth to very high bandwidth codecs in ST 8.1 between sites, with no change at all. Also, the conf bridge is a 7.1 bridge, which no longer has the VOIP Setting tab. I have been assured by ST TAC that the setting is properly set and will not cause any issues. Still no luck though...
Thanks for the responses guys! Very appreciated!
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Old 04-07-2009, 01:05 PM   #6
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How are the numbers coming into the conference bridge? PRI?
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Old 04-08-2009, 04:00 PM   #7
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Quote:
Originally Posted by benchtoplabs View Post
On the conference bridge, look under the "Configuration" tab
then "VOIP Settings" What do you have in these boxes?
Great question not answered yet and most likely the key to the issue.

I will add

Is SIP enabled on the ShoreTel System?

If Yes then you need to follow the CB Config note FAQ very carefully for the proper Ports to be used as per the question in the quote
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