[Spce-user] 400 Normal Release

[Dave Massey]

You guys are probably going to get sick of me, but so far my voip experience isnt starting off great.

I have 2 peers, one is voxcentral and one is TieUS, I only went with the 2nd one because they can port numbers in a particular rural area (519587)

Voxcentral works OK but TieUS if Im making a call out from the subscriber to the PSTN the audio from the peer stops 10 seconds in and I get the below:
There is no BYE and the audio is sent from the subscriber the whole time, I only get 2 way audio for 10 seconds.
On a side note, also no DTMF tones pass from the PSTN end back to the subscriber.  Ive tried 2 different makes of ATAs.


04:26:55.688459 IP (tos 0x0, ttl 118, id 50699, offset 0, flags [none], proto UDP (17), length 615)
    209.139.240.87.5060 > 24.102.50.52.5060: SIP, length: 587
	SIP/2.0 400 Normal Release
	Via: SIP/2.0/UDP 24.102.50.52;branch=z9hG4bKd58f.25ac20bcb2bd9393dcaf7053c0f73282.0
	Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bKHeu3Eaoy;rport=5080
	From: <sip:15198053004 at sip.optionsdsl.ca>;tag=3A26561C-507F4CC1000DC537-CE5AF700
	To: <sip:19057722572 at 209.139.240.87>;tag=GR52RWG346-34
	Call-ID: b30e737d at 10.40.36.124_b2b-1
	CSeq: 10 INVITE
	Contact: "Verso CM" <sip:19057722572 at 209.139.240.87:5060>
	Expires: 60
	P-Asserted-Identity: <sip:15198053004 at sip.optionsdsl.ca>
	Allow: ACK, NOTIFY, OPTIONS, REFER, INFO, BYE, CANCEL, INVITE
	Content-Length: 0



If I call INward towards the subscriber from the PSTN it works and the audio doesnt stop. (Still no DTMF though)
I have no idea if this is the peer problem or something Ive not set up right?