As CME also needs the SIP service I would recommend to put the bind statements for the ITSP service in either a tenant or on the dial peers facing the service provider instead of setting it in global configuration.
Yes one file for the output from the debugs please.
Why do you need to be on the site to restart? Just do a restart command in the cli.
For the fax part of your question, let's focus on one thing at a time. When you get your current issue resolved we can deal with the next thing. No point in discussing other things at this point.
Hello Roger,
Thank you very much for help. i add conf that you send me and reload device.
I can not make call from outside( on my mobile phone i can not hear anything just silence). Debug is attached in one file (debug voip ccapi inout and debug ccsip message). I hope that i sent correctly.
Conf which i add is:
voice service voip
ip address trusted list
ipv4 0.0.0.0 0.0.0.0
address-hiding
mode border-element
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
no supplementary-service sip refer
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
sip
bind control source-interface GigabitEthernet0/0/0
bind media source-interface GigabitEthernet0/0/0
registrar server
no call service stop
voice class uri PSTN sip
host ipv4:10.10.38.5
voice class codec 10
codec preference 1 g711alaw
codec preference 2 g711ulaw
codec preference 3 g722-64
codec preference 4 g729r8
voice translation-rule 100
rule 1 /^678$/ /+38736325353/
voice translation-rule 200
rule 1 /^\+38736325353$/ /678/
voice translation-profile Dolazni-HT
translate called 200
voice translation-profile Odlazni-HT
translate calling 100
dial-peer voice 200 voip
corlist outgoing Test
translation-profile outgoing Odlazni-HT
destination-pattern 0T
session protocol sipv2
session target ipv4:10.10.38.5
voice-class codec 10
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
incoming uri via PSTN
voice-class codec 10
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
i have problem to upload plain text here.
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.
I see you have removed all your allow connection lines. That will break the CME service for all calls, ie from an internal extension to another internal extension and from/to an internal extension to/from any external number. You need to have "allow-connections sip to sip", otherwise no calls will work.
Please share the running configuration as it looks now after the updates you made and also the IP address of the phone device that is receiving the call, plus the output from a show ip route in the router.
