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SIP invite to voiceflex

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gmacdonald78

Programmer
May 4, 2012
49
GB
Odd one here system being in and running fine for months with no issues, all of a sudden outgoing calls fail on the SIP trunk. Looking at the traces from the IPO and voiceflex the invite header is wrong:

08/07/16 10:44:46-492ms Line = 17, Channel = 1, SIP Message = Invite, Call Ref = 95, Direction = From Switch, From = 0127xxx990@sip28.voiceflex.com, To = 07708744867sip28.voiceflex.com@sip28.voiceflex.com
08/07/16 10:44:46-513ms Line = 17, Channel = 1, SIP Message = Response, Call Ref = 95, Direction = To Switch, From = 01277xxx990@sip28.voiceflex.com, To = 077xxx44867sip28.voiceflex.com@sip28.voiceflex.com, Response = 401 Unauthorized

sip28.voiceflex twice?????
 
Your dial shortcode is probably incorrect.

You'll see in monitor what's dialed and what it adds.

"Trying is the first step to failure..." - Homer
 
Odd one here system being in and running fine for months with no issues, all of a sudden outgoing calls fail on the SIP trunk
So what has been changed
What hae Voiceflex changed?
What has the customers IT Dept. changed on the network?


Do things on the cheap & it will cost you dear
 
You're missing the @ in the first instance, but forget that, this will be bad shortcode programming I'd wager, bearing in mind you don't need to add the "@sip.co.uk" part into shortcodes anymore just the number as per analogue/ISDN trunks :)

 
Bang on re-built the SIP trunk and ARS and noticed that the original one was missing the @ before the ITSP address, what's cooking my noodle now is how was this working for the last 6 months?????????????
Thanks for your help.
 
As mentioned you don't need the @blahblah.com part in ARS/shortcodes now :)

 
Really? So what should the ARS short code have if not the @blahblah.com? Clearly I'm failing to keep up
 
?
Dial
.
Line group

And cosidering it suddenly stopped working something must have happened, the way it's sending out To Field now should never have worked.

"Trying is the first step to failure..." - Homer
 
If it was:

Code 0N
Feature Dial
Tel number 0N"@blahblah.com"
Line gp - SIP trunk

now just

Code 0N
Feature Dial
Tel number 0N
Line gp - SIP trunk



 
Odd ive just had a second site that's had sip trunks with voiceflex for years without any issues and on Thursday they stopped being able to make out bound calls. Added @ to the ARS and calls are routing out fine.....leads to think voiceflex must have changed something?
 
Don't know which IPO version you're using but if the SIP Line is configured correctly you shouldn't need to add @ in ARS as it will break some call scenarios.

"Trying is the first step to failure..." - Homer
 
Voiceflex definitely changed something last week as we had a customer who moved from username authentication to IP authentication about 18 months ago. Oddly no one removed the username/password from the SIP trunk setup so it was doing both for 18 months. Last week the trunk failed due to it trying to register whilst being IP authenticated.

| ACSS SME |
 
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