Is the branch sending the 9 when it comes over on trunk 18?
Shouldn't you actually have 9N / Dial / N / 17 go instead to an ARS table that routes through trunk 17?

Thanks for the reply.

1. Yes, it sends 9 along with the actual PSTN number
2. Sorry Im a Cisco guy and have no idea about the ARS table. Please suggest me a simplest way to achieve this.

How do you dial out of the home system? Dial 9 and number? Dial something else and the number? Just dial the number without any leading prefix?

I'm US based, so I can explain the scenario used here.

Users on the Main system dial 9 and telephone number to make an outgoing call.
Short Code 9N / Dial / N / 50:Main strips off the 9 and sends the telephone number to ARS table 50:Main
ARS table 50:Main has some pattern matching, but basically the entry N; / Dial 3K1 / N / {Outgoing line group (17)}

Similar thing at the Branch system
Short Code 9N / Dial / N / ARS 50:Main
ARS 50:Main N; / Dial 3K1 / 9N / Line Group 18

So the Branch User dials 9-1-800-555-1212. 18005551212 is sent to 50:Main (on the Branch system). 918005551212 is sent to line group 18, which comes over to the Main system. Main system receives 918005551212 and sends 18005551212 to 50:Main (on the Main system) sends 18005551212 on the SIP trunk 17

SIP (and PRI) dialing is done "en-bloc", all digits are accumulated until you are finished dialing them on the phone, and then they are sent out on the line.

Yes we do dial in the same way. 9+PSTN number for the local calling.

I have done following configurations:
1. Incoming Call Route: Line No 18, Destination "." , everything else default.
2. Short code created with 9N/Dial/N/50:Main
3. ARS 50: Main has following entries
a. ?/./Dial/0
b. N/N/Dial 3K1/17

when call comes in from Branch, Line 18 trace shows error "404 Not Found", No activity recorded on Line 17

I think perhaps entry a is overriding entry b

Did you try to enable Call routing method "TO Header? And do the users from system b send a valid URI?

SIP messages from system a would be helpful.

Need some help with IP Office? https://www.fwilke.com/home
IP Office SIP trunk configuration: https://www.fwilke.com/en/ip-office-sip-trunk-en/
CLI based cale blocking: https://blog.fwilke.com/ip-office-block-calls-from...
SCN fallback over PSTN: https://blog.fwilke.com/ip-office-scn-fallback/

TTT
I have tried removing the a) a. ?/./Dial/0, doesnot make any difference.
Derfloh

Did you try to enable Call routing method "TO Header? And do the users from system b send a valid URI? Yes, the routing method is Header.

Here is the SIP trace from the Line 18, nothing on Line 17

9/7/19 2:36:20 PM-959ms Line = 18, Channel = 1, SIP Message = Response, Direction = From Switch, From = XXXXX500@192.168.102.21, To = 955971111@192.168.10.10, Response = 100 Trying
9/7/19 2:36:20 PM-962ms Line = 18, Channel = 1, SIP Message = Response, Direction = From Switch, From = XXXXX500@192.168.102.21, To = 955971111@192.168.10.10, Response = 404 Not Found
9/7/19 2:36:21 PM-197ms Line = 18, Channel = 1, SIP Message = Ack, Direction = To Switch, From = XXXXX500@192.168.102.21, To = 955971111@192.168.10.10

We need a Monitor trace.

Need some help with IP Office? https://www.fwilke.com/home
IP Office SIP trunk configuration: https://www.fwilke.com/en/ip-office-sip-trunk-en/
CLI based cale blocking: https://blog.fwilke.com/ip-office-block-calls-from...
SCN fallback over PSTN: https://blog.fwilke.com/ip-office-scn-fallback/

I have following message from the Monitor:

Reason: Q.850;cause=1;text="Unallocated (unassigned) number"

I have been struggling to achieve this pretty simple looking task from last 3 weeks.Please assist.

As derfloh said you need to post a complete monitor trace to see exactly what is going on. Please post in a .txt file and attach it rather then post it on here. Just giving 3 lines here and 1 line here is impossible to problem solve.

The more information you provide the better on here.

The truth is just an excuse for lack of imagination.