[Spce-user] 400 Normal Release

[Dave Massey]

OK I did a test call, and attached the output.  
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as for the DTMF, Im not officially yet calling a 519587 number, just a test DID they have set up (Im not sure that if they had the PRI it even comes into play yet for this DID?)  I did try the test anyways, and I called from a PSTN line that is name/number capable  to the 519 DID, and I did not get the caller name, just the number.  
This could really suck, as they were the only provider that I know of on the planet that could port 519587 numbers.


On 2012-10-18, at 4:56 AM, Skyler <skchopperguy at gmail.com> wrote:

> 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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> 
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