[cisco-voip] SIP gateway config

Ryan Ratliff rratliff at cisco.com
Mon Apr 26 17:18:51 EDT 2010


Only if you need SIP early media from CUCM.

-Ryan

On Apr 26, 2010, at 5:06 PM, miken miken wrote:

> MTP configured and box checked mandatory in SIP trunk configuration on CUCM?
> 
> Thanks
> MikeN
> 
> On Mon, Apr 26, 2010 at 3:00 PM, Brian Schultz <bms314 at gmail.com> wrote:
> Yep, have that already.  Gig0/1.110 has the IP address configured on the SIP trunk in CUCM. 
>  
> voice service voip
>  sip
>   bind control source-interface GigabitEthernet0/1.110
>   bind media source-interface GigabitEthernet0/1.110
>  
> I also have the following:
>  
> voice class codec 1
>  codec preference 1 g711ulaw
>  codec preference 2 g729r8
> dial-peer voice 100 voip
>  destination-pattern ....
>  session protocol sipv2
>  session target ipv4:172.21.20.10
>  voice-class codec 1
>  dtmf-relay rtp-nte
>  no vad
> dial-peer voice 1 pots
>  incoming called-number .
>  direct-inward-dial
> 
> 
>  
> On Mon, Apr 26, 2010 at 3:58 PM, Ahmed Elnagar <ahmed_elnagar at rayacorp.com> wrote:
> You have to bind signal and media traffic out of the router to the CUCM with the IP address you have configured on CUCM “by default CUCM reject calls with source address other than the configured”
> 
>  
> Try the below on the gateway:
> 
>  
> voice service voip
> 
> sip
> 
> bind all source-interface “interface configured on CUCM”
> 
>  
>  
>  
>  Best Regards;
> 
>   Ahmed Elnagar
> 
>   Senior Network PS Engineer
> 
>   Mob: +2019-0016211
> 
>  
> 
>  
> From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Brian Schultz
> Sent: Monday, April 26, 2010 10:34 PM
> To: cisco-voip at puck.nether.net
> Subject: [cisco-voip] SIP gateway config
> 
>  
> Does anyone happen to have an example SIP gateway config for an ISR?  CUCM 8.0(2) with a SIP trunk to a 2921 gateway (15.0.M1.12) with a standard PRI for PSTN access.  I have outbound working, but inbound gives a fast busy with a 401 Unauthorized in the SIP debug.
> 
>  
> Thanks,
> 
> Brian
> 
>  
>  
>  
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