09-03-2015 08:46 PM - edited 03-17-2019 04:13 AM
Now I have testing sip srst 3905 and SCCP 7962 when cucm down its register with CME but 3905 can't call to 7962 but 7962 can call 3905 when I off-hook I got busy tone please see config below
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 t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
sip
registrar server expires max 600 min 60
!
voice class codec 1
codec preference 1 g711alaw
codec preference 2 g711ulaw
codec preference 3 g729r8
!
!
!
voice register global
mode srst
system message srst active
max-dn 100
max-pool 100
!
voice register pool 1
id network 192.168.100.0 mask 255.255.255.0
dtmf-relay rtp-nte cisco-rtp sip-notify
no vad
At the cucm(192.168.100.102) srst reference I config IP address and Sip network already advise me if anything is missing ....
Solved! Go to Solution.
09-04-2015 11:14 AM
Hi,
You are missing bind command and source interface.
voice service voip
bind all source x/x
!
voice register global
source-address x.x.x.x
Please add these and test. If things didn't work, please share debug ccsip mess
09-03-2015 10:31 PM
Please try configuring the following:
voice service voip
ip address trusted list
ipv4 192.168.100.0 255.255.255.0
If that does not work enable "voice iec syslog" (in global config) and debug ccsip messages (ensure you log buffer is large enough to capture everything) when in SRST, then reproduce the 3905 busy issue.
Once you do this can you please post a copy of the "show log command"?
Additionally, what version of IOS are you running?
09-04-2015 09:22 AM
Hello jonathan
My version IOS is 15.5.let me test I will update again
09-04-2015 11:14 AM
Hi,
You are missing bind command and source interface.
voice service voip
bind all source x/x
!
voice register global
source-address x.x.x.x
Please add these and test. If things didn't work, please share debug ccsip mess
09-07-2015 05:35 AM
Hi Natt,
Did you apply the bind command what is the results?
09-07-2015 01:53 AM
under the sip, try to add:
bind control source-interface <your srst interface>
bind media source-interface <your srst interface>
and then in the voice register global apply the command:
create profile
09-09-2015 11:27 PM
Discover and save your favorite ideas. Come back to expert answers, step-by-step guides, recent topics, and more.
New here? Get started with these tips. How to use Community New member guide