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SRST with SIP phones 8961 on Cisco 2921.

Hello All,

I configured Cisco 2921 as SRST router and H323 gateway. My phones (Cisco 8961) are registered.

I cannot call in from PSTN and pickup the call, but for some reason I cannot dial any extension or PSTN number.

Config and show commands in attachment.

When I dial any number seems like a phone cannot even dial it, debug shows nothing (ccapi inout)

7 REPLIES 7
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Cisco Employee

Hi Salnikov,

What is the behavior of the phone? What happens when you dial a number on the SIP phone? Do you get re-order tone?

What debugs did you enable on the CME?

What's the number you are trying to call?

What do you see if you enable debug voip ccapi inout?

Thanks

Sree

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Hi Sree,

Sure I'll give you more details.

Phone shows SRST MODE system message.

When I dial any number (extension or Pstn) looks like a phone waits for digits, but after luke 10 seconds it sound like fast busy tone (if I am not wrong it's reorder tone that you mentioned).

I debuged voice ccapi inout. It doesn't show anything.

I am trying to call ext 3xx and 9xxxxxxxx numbers.

Looks very strange bacause I assume I should be able to place at least internal call...

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Hi Salnikov,

Have you bind your sip to one of your active interface ?

bind control source-interface ???

bind media source-interface ???

Rgds

Novri

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Hi,  Novri,

No, I added bind all source-interface ge0/0

But currently I have only one physical interface and IP (I replaced trunk by access).

This interface has connands

h323-gateway voip interface

h323-gateway voip bind srcaddr 10.13.206.2

So, use one interface for bind h323 and SIP. Is it acceptable or I should create Loopback interface for the SIP binding? Should it be routed to phone subnet?

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Hi Salnikov,

You can create loopback interface and bind the sip to that interface, the ip address must be routeable to phone subnet.

Rgds

Novri

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Novri and Carl,

Thank you for your advices, I will arrange with customer, test and give you my answer.

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Participant

Agreed with Novri. I always bind the loop back interface to sip I find without this it won't work.

Thanks, Carl Ratcliffe


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