[Spce-user] 400 Normal Release

[Skyler]

Hi,

  For the first issue, a time-out is occurring. I'd say to run 
'ngrep-sip b > /tmp/tieus-out-audiodrop' and make an outbound call from 
subscriber over that trunk. Then post contents here.

  On the second issue, DTMF is probably out-of-band ss7 and not reaching 
or not passed through TieUS. 519587 shows Bell as the ILEC, there could 
be many reasons DTMF is not coming through. Sounds like a tieus ticket 
in yer future, could take weeks to figure that one out; if at all. A 
quick test, use a PSTN phone that you know always shows CLI-Name + 
number then call your 519587xxxx subscriber. Do you see the name? if 
not, then TieUS has a PRI using MF (in-band) and will never get the DTMF 
from ss7. If you do, then its a config issue on their switch.

--Skyler

On 10/17/2012 5:35 PM, Dave Massey wrote:
> 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?
>
>
>
>
>
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