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New Member

CUBE SIP Trunk dial-peer

Hi,

I'm using my Cisco 2821 to act as a CUBE.

I have configured the below:-

!
voice service voip
 allow-connections h323 to h323
 allow-connections h323 to sip
 allow-connections sip to h323
 allow-connections sip to sip
!
sip-ua
 credentials username XXXXX password XXXXX realm XXXXXX.XXX.XXX
 authentication username XXXXX password XXXXX realm XXXXXX.XXX.XXX
 retry invite 3
 retry response 3
 retry bye 3
 retry cancel 3
 retry register 3
 retry options 10
 timers trying 1000
 timers connect 1000
 timers register 1000
 registrar ipv4:X.X.X.X expires 3600
!
dial-peer voice 316 voip
 destination-pattern 316........
 session protocol sipv2
 session target ipv4:X.X.X.X
 session transport udp
 incoming called-number .
 dtmf-relay sip-notify
 no vad
!

After the configuration, i do a show sip-ua register status.

I see all my pots dial-peer try to register, which they are not suppose to. I stops this by issuing the command "no sip register".

I don't see any SIP voip dial-peer trying to register, the above dial-peer voice 316 voip also does not try to registers.

Anyone can help point out to me on why my pots dial-peer tries to register but not my voip dial-peers ?

 

Thanks.

5 REPLIES

Usage Guidelines"SIP gateways

Usage Guidelines

"SIP gateways can register E.164 numbers on behalf of analog telephone voice ports (FXS), IP phone virtual voice ports (EFXS), and SCCP phones with an external SIP proxy or SIP registrar. The command show sip-ua register statusis only for outbound registration, so if there are no SCCP phones or FXS dialpeers to register, there is no output when the command is run."

 

Thanks

Manish

New Member

If i can't use the show sip

If i can't use the show sip-ua register status to check the whether the SIP trunk from the CUBE has successfully established the trunk with the service provider using the credentials i have entered.

What command or how do i debug this..

VIP Super Bronze

Its not all providers that

Its not all providers that require you to register before you can use the service. Have you tried making a call? Does the call succeed or does it fail?

Please rate all useful posts "The essence of christianity is not the enthronement but the obliteration of self --William Barclay"
New Member

The call fails.. And

The call fails.. And according to the service provider, they need to see the SIP trunk up on their side 1st before any calls can pass thru.

Please activate a debug ccsip

Please activate a debug ccsip message an try to make a call.

After that, please post the output.

 

 

Thanks

 

Regards

 

Carlo

Please rate all helpful posts "The more you help the more you learn"
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