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SES with 3'rd party SIP endpoints

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saifbarry

MIS
Joined
Sep 4, 2012
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2
Hello,

I'm copying and pasting this from another forum as it very accurately describes the exact problem I'm facing. Unfortunately, the OP never reverted if he actually solved the issue.

------------------[ Begin ]-----------------------------
I have trying to setup a SES server, to get some SIP dect phones working.

I have installed the 5.2.1 applied the sp1 and to my best knowlegde set it up correctly.

I'm running a ACM 5.x.

I have created the endpoints in the SIP server and "attached" an CM endpoint number.
I have created a SIP trunk, and the stations in ACM with a OPS offstation feature.

I can dial from the SIP endpoint to an outside line voice both ways.
I can dial from the outside to the SIP endpoint voice both ways.
I can dial a H.323 station from the SIP endpoint with voice both ways.

I cannot dial the SIP endpoint from the H323. station. The sip endpoint rings but when I try to pickup the call I get a new dial tone.
And I can hear the call keep dialing in the H.323 endpoint.

What have I done incorrectly?
Here are a trace of the H.323 station (20394) dialing the SIP endpoint(20391)
14:46:57 active station 20394 cid 0x13a
14:46:57 G711A ssff ps:20
rgn:1 [10.15.1.53]:2484
rgn:1 [10.15.1.10]:2050
14:46:58 dial 20391
14:46:58 term station 20391 cid 0x13a
VOIP data from: [10.15.1.10]:2058
14:47:08 Jitter:0 0 0 0 0 0 0 0 0 0: Buff:8 WC:0 Avg:0
14:47:08 Pkloss:0 0 0 0 0 0 0 0 0 0: Oofo:0 WC:0 Avg:0
VOIP data from: [10.15.1.10]:2050
14:47:08 Jitter:0 0 0 0 0 0 0 0 0 0: Buff:8 WC:1 Avg:0
14:47:08 Pkloss:0 0 0 0 0 0 0 0 0 0: Oofo:0 WC:0 Avg:0
14:47:15 idle station 20394 cid 0x13a

And a trace of the Trunk during the call.
14:49:12 dial 20391
14:49:12 seize trunk-group 4 member 13 cid 0x141
14:49:12 Calling Number & Name NO-CPNumber NO-CPName
14:49:12 Setup digits 20391
14:49:12 Calling Number & Name 20394 Thomas S,Laur
14:49:12 Proceed trunk-group 4 member 13 cid 0x141
14:49:13 Alert trunk-group 4 member 13 cid 0x141
14:49:15 G711A ssff ps:20
rgn:1 [10.15.1.41]:58000
rgn:1 [10.15.1.10]:2056
14:49:15 xoip options: faxff modemff tty:US uid:0x5005f
xoip ip: [10.15.1.10]:2056
14:49:25 idle trunk-group 4 cid 0x141
------------------[ End ]-----------------------------

Anyone face anything like this?
 
Upon pickup of the sip endpoint, it is actually a stutter dial tone.
And if we input literally any digit during this dial tone, the call completes, as in we can now talk to the calling party.

Does this ring any bells with anyone?
 
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