Help on Routing Calls with SIP

[Agent0007]

Hello Guys!

I need help on how to route calls with SIP via Flex DID Conv, i have it configured to route it using the flex did conv table but it does not work.

with normal CO Calls it works but with SIP it does not, does anyone has had the same issue before?

Thank you!

Best regards,
RR

 

[martinhodge]

what system is this on . and what sip provider are you using .

 

[Agent0007]

the system is eMG80 and the provider is Sipgate, it cannot route calls automatically, if i manually route call using the desktop phone it works, it also works with preset call forward, but if I use any conversion table it just does not work, it does not route calls to other stations or mailboxes.

 

[nfcphoneman]

Things to check:

1) In PGM 140, verify that your CO Type is set to "DID"

2) In 145, verify that "DID Start Signal" is set to "Immediate"

"DID Conversion Type" is set to "Modify Using Flexible DID Conversion Table".​

"Number of Digits Expected from DID Circuit should be 3 or 4.​

"DID Digit Mask" = #*** (if expecting 3 digits) or **** (if expecting 4 digits)​

3) In PGM 126, verify that "Ring Route Type" = DID CONVERSION

"DID Conversion Type" = Modify Using Flexible DID Conversion Table​

"Number of Digits Expected from DID Circuit should be 3 or 4.​

"DID Digit Mask" = #*** (if expecting 3 digits) or **** (if expecting 4 digits)​

4) In PGM 231, make sure to program the last 3 (or 4) digits of your DID's to route to the proper place.

http://www.jacksonvilletelephone.com

Jacksonville Florida Telephone Systems

 

[Agent0007]

@nfcphoneman

Thank you but I already tried that config and it did not work. I figured out another way by using CO/IP Ring Assignment(114)

I forwarded it to a hunt group that forwards it to the VSF Group at night time, because a direct forward to VSF just won't work​

and I still don't know why. It just does not follow the conversion tables. Thank you for your response!!​

 

[xs650]

Isn´t it needed to cut the leading digits in DID Remove Number to get the DID?
When I got 1234567 00-99 from the provider, I have to cut 7 digits to get the
DID 00-99, or I am wrong?

 

[nfcphoneman]

Are you trying to route using 3 digits or 4?

http://www.jacksonvilletelephone.com

Jacksonville Florida Telephone Systems

 

[nfcphoneman]

xs650, I don't believe you have to cut any digits. You just need to tell the system if you're routing using 3 or 4 digits.

http://www.jacksonvilletelephone.com

Jacksonville Florida Telephone Systems

 

[xs650]

long time ago but I think I have cut them cause I have routed different DID length
(-0 and 100-199) via flex did conversion table.

 

[martinhodge]

do a wire shark and see what numbers are being sent from your sip provider .

this way you can see what is actually being sent and if they are sending the correct number .

 

[Thermus]

Agent0007
How does the SIP call get to the desktop phone so you can manually route it?
Is it routing via PGM 144 to get there?

It sounds like the SIP lines are either set to CO in PGM140, or you are not receiving the digits you are supposed to be receiving.

Another option is to change PGM 126 "ring route type" for the number you expect to receive, to "ID Assigned station"

Then Assign that SIP user ID to the extn that needs to receive the call in PGM111 "SIP USER TABLE INDEX"

But as suggested above, either wireshark or do a SIP Message trace on the phone system to see what you are being sent