06-28-2018 09:25 AM - edited 03-17-2019 01:07 PM
Hello,
We have a number of new DNs which we've added to CUCM. They are all in London so begin 0203 which is new to our company so I updated the Gateway dial-peers to suit.
We CAN dial out from the test number but are UNABLE to dial IN
PSTN >>> SIP PROVIDER >>> GATEWAY >>> CUCM
debugs collected when dialing in are attached .txt
Can anyone assist me with this issue please?
Solved! Go to Solution.
07-04-2018 09:34 AM
Hello,
I've reviewed the Gateway config and realised the problem was with the incoming number specified on dial-peer voice 9
I've now changed to incoming called-number ^0[1-2].........$
Calls into 01...... and 02..... number are now working
Thank you for your help and assistance on this issue.
06-28-2018 10:27 AM
06-28-2018 11:16 AM
What is the exact issue you face? Like what is the behavior of the call?
From signalling perspective, it is fine till 200 OK message received on the gateway but nothing after that.
06-29-2018 12:30 AM - edited 06-29-2018 12:31 AM
Thank you both,
The call in to any 0203 number just hangs with no audible feedback (i.e. like busy or reorder tone)
Its obviously getting to the Gateway but I don't think CUCM receives the call (I've checked using RTMT call logs).
06-29-2018 02:46 AM
06-29-2018 06:25 AM
Hello,
I've now attached the full debug I collected the other evening (I can only collect out of hours as the gateways are remote to me and debugs would lock me out during the day).
If these aren't suitable I will re-collect over the weekend and update the ticket.
Let me know. Thank you.
06-29-2018 09:57 AM
07-02-2018 12:51 AM
Calling: +447534123456
Called: +442037654321
07-02-2018 09:17 AM
07-03-2018 02:56 AM
07-03-2018 09:17 AM
07-04-2018 09:34 AM
Hello,
I've reviewed the Gateway config and realised the problem was with the incoming number specified on dial-peer voice 9
I've now changed to incoming called-number ^0[1-2].........$
Calls into 01...... and 02..... number are now working
Thank you for your help and assistance on this issue.
07-04-2018 10:57 AM
07-05-2018 01:31 AM
Compare the following:
!
dial-peer voice 9 voip
description *** Inbound DP from ITSP ***
call-block translation-profile incoming call_block
call-block disconnect-cause incoming call-reject
session protocol sipv2
incoming called-number ^01.........$
voice-class codec 1
voice-class sip bind control source-interface GigabitEthernet0/1
voice-class sip bind media source-interface GigabitEthernet0/1
dtmf-relay rtp-nte
ip qos dscp cs3 signaling
no vad
!
changed to....
!
dial-peer voice 9 voip
description *** Inbound DP from ITSP ***
call-block translation-profile incoming call_block
call-block disconnect-cause incoming call-reject
session protocol sipv2
incoming called-number ^0[1-2].........$
voice-class codec 1
voice-class sip bind control source-interface GigabitEthernet0/1
voice-class sip bind media source-interface GigabitEthernet0/1
dtmf-relay rtp-nte
ip qos dscp cs3 signaling
no vad
!
07-05-2018 07:57 AM
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