Accepted Solutions
Absolute sure that I did not suggest you to add this.
ip address trusted list
ipv4 0.0.0.0 0.0.0.0
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323Please correct and then get back with the debug in a text file. I don't know why you can't upload a plain text file, that normally works. What text editor do you use, suggest that you use Notepad++ and save it as plain text file.
With all the debugs you have turned on it makes it harder to pinpoint the cause of your issue as there is so much white noise in the output. Can you please redo the call with these debugs running, debug ccsip message and debug voip ccapi inout, also please share your running configuration so that we can look it through.
I haven’t checked your TAR file and likely will not as I’m kind of restrictive in opening files other than just plain text. Based on the information provided about the dial peers I will suggest that you remove these two lines from your inbound dial peer.
session target ipv4:10.10.38.5:5060
incoming called-number .%
The first line is used on outbound dial peers, as such it’s not needed on an inbound dial peer. The second line is not a good way to match inbound calls, replace this with a match that uses information in the VIA header. See this document for how to, In Depth Explanation of Cisco IOS and IOS-XE Call Routing
I see you deleted what I suggested, but you have not added what I suggested. Without any configuration to match your inbound dial peer it won’t work.
Please have both of the debugs running and post the output. That’s what I wrote in my first reply.
Also please post the output in a txt file, not a docx.
Thanks for the text files, I will look at the configuration. However it's now the third time I ask you to provide the debugs in one file. I'm sorry, but I will not look at individual files with debug output as you'd want to see the output of these in combination.
Try with adding these changes.
voice service voip
ip address trusted list
no ipv4 0.0.0.0 0.0.0.0 !Not a good thing to have as it opens up for toll fraud, you'd want to have the ITSP SIP server IP(s) defined
ipv4 <ITSP SIP server IP>
! add as many lines as you need
no allow-connections h323 to h323 !Not needed with SIP
no allow-connections h323 to sip !Not needed with SIP
no allow-connections sip to h323 !Not needed with SIP
address-hiding
mode border-element !Save and reboot once you have added this and the rest of the suggested changes to activate Cube
sip
bind control source-interface GigabitEthernet0/0/0
bind media source-interface GigabitEthernet0/0/0
!
no voice class codec 10 ! You don't need all those codecs
voice class codec 10
codec preference 1 g711alaw
codec preference 2 g711ulaw
codec preference 3 g722-64
codec preference 4 g729r8
!
no voice class sip-profiles 1 !Not needed when you have turned on Cube functionallity with mode border-element and have address-hiding in global config
!
voice translation-rule 100
rule 1 /^678$/ /+38736325353/ !Set to match starts and ends with, ie will only match the exact string
!
voice translation-rule 200
rule 1 /^\+38736325353$/ /678/ !Set to match starts and ends with, ie will only match the exact string and also add start with a plus (+)
no rule 2 !Not needed as it is the same number as rule 1
!
voice class uri PSTN sip !Defines what to look for the the SIP header, used to match the IP in the VIA header on the inbound dial peer 201
host ipv4:10.10.38.5
!
dial-peer voice 200 voip
corlist outgoing Test
translation-profile outgoing Odlazni-HT
destination-pattern 0T !No need to have a dot after the zero and before the T
session protocol sipv2
session target ipv4:10.10.38.5:5060
voice-class codec 10
no voice-class sip profiles 1
voice-class sip bind control source-interface GigabitEthernet0/0/2
voice-class sip bind media source-interface GigabitEthernet0/0/2
dtmf-relay rtp-nte sip-notify
no vad
!
dial-peer voice 201 voip
translation-profile incoming Dolazni-HT
session protocol sipv2
no session target ipv4:10.10.38.5:5060 !Used on outbound dial peers
no incoming called-number .% !Not a good way to match inbound dial peer
incoming uri via PSTN !Matches inbound dial peer based on information in VIA header
voice-class codec 10
no voice-class sip profiles 1
voice-class sip bind control source-interface GigabitEthernet0/0/2
voice-class sip bind media source-interface GigabitEthernet0/0/2
dtmf-relay rtp-nte sip-notify
no vad
