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SIP SRST 3905 issue please help

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 ....

1 Accepted Solution

Accepted Solutions

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

 

View solution in original post

6 Replies 6

Jonathan Unger
Level 7
Level 7

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?

Hello jonathan

    My version IOS is 15.5.let me test I will update again

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

 

Hi Natt,

 

Did you apply the bind command what is the results?

Laith Ibrahim
Level 1
Level 1

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

 

 

Hi Mohammad

    That is work

 

Thanks,

Nattapong