Hi Gents,
I have a setup of CUCM---SIP Trunk--->CUBE(ISR4331)---SIP Link--->SIP Provider.
provider asks to change the Min-se and and session expiry to the following values: min-se 90 session-expires 300, so I applied sip profiles to tune these values to the default CUCM values of 1800.
everything works fine outbound and inbound calls, only one issue that outbound calls disconnects after around 5 minutes, debug output shows a BYE message received from SIP provider side which cause the call to be dropped.
When opened a case with provider they said the call is dropped because our cube "is not sending the Session Timer update message during the call".
I configured session refresher under SIP global config and also under dial-peers with no success.
I assume that CUCM is by default sending the re-invite on have expiry timer which is 1800 (30 minutes) so every 15 minutes according to its own timers, which will not help in our case as the session expiry timers with the provider is only 300 seconds, this is only an assumption, so if the CUBE send the re-invite at it's configured values this should fix the issue (assumption).
Note: Media Termination Point Required checkbox is checked on the CUCM SIP trunk pointing to the CUBE.
here is related config:
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
fax protocol none
h323
sip
rel1xx disable
min-se 90 session-expires 300
session refresh
registrar server expires max 600 min 60 %%used for SRST for SIP phones (should not be related to the issue)
no update-callerid
early-offer forced
midcall-signaling passthru
no call service stop
voice class sip-profiles 1
request INVITE sip-header Min-SE modify "90" "1800"
request INVITE sip-header Session-Expires modify "300" "1800"
!
voice class sip-profiles 2
request INVITE sip-header Min-SE modify "1800" "90"
request INVITE sip-header Session-Expires modify "1800" "300"
dial-peer voice 10 voip
description **<<Tunning-DTMF-For-Incoming-Calls>>**
translation-profile incoming INCOMING-CALLS
session protocol sipv2
session transport udp
incoming called-number 17762222
voice-class codec 10
voice-class sip session refresh
dtmf-relay rtp-nte h245-signal h245-alphanumeric
no vad
!
dial-peer voice 20 voip
description **<<OUTGOING-Calls>>**
translation-profile outgoing OUTGOING-CALLS
destination-pattern 9.T
session protocol sipv2
session target ipv4:10.197.0.193:5060
session transport udp
incoming called-number 9.T
voice-class codec 10
voice-class sip profiles 2
voice-class sip session refresh
voice-class sip audio forced
dtmf-relay sip-notify sip-kpml rtp-nte h245-signal h245-alphanumeric cisco-rtp
no vad
So how could I fix this issue and make the CUBE send the update or re-invite message during the call to avoid call dropping?
attached is debug output for debug ccsip message/states/call/event for a call from extension 5290 to 937151314
@Ayodeji Okanlawon
