cme sip phones unable to call outside

[antonello.moneta]

Hello,

I used this configuration to register a sip phone:

!--enable server registration
voice service voip
sip
registrar server
exit
exit
!-configure register
voice register global
mode cme
source-address 172.xx.xx.xx port 5060
max-dn 3
max-pool 3
tftp-path flash:
create profile
exit
!--config number
voice register dn 1
number 344
allow watch
name SIP-Client
exit
!--Config pool for sip phone
voice register pool 1
id mac 0011.9566.7535
number 1 dn 1
username test password test
codec g711ulaw

The sip client is correctly registred, it can call any internal ext and I  can call it from internal ext, but I cannot call outside neither pstn  and sip trunk.

Any suggestion ?

Thank you really much for your help.

[Adam Thompson]

Assuming that your outbound dial-peers are setup correctly, try adding the bind statement under voice service voip.

Example:

voice service voip

  sip

    bind media source-address loopback 0

    bind control source-address loopback 0

HTH

-Adam

Please rate posts if they are helpful

[antonello.moneta]

Hi Adam,

Thanks for your help.

The dial-peer are ok, since I can call out form sccp phones. Only  sipphones cannot call out.

I tried your tip, but it still not working.

I think I miss something on the configuration, that not let me go out form sipphones, but I don't know what.

Any other suggestion.

[Arun Kumar]

Are you able to receive external calls (from PSTN or SIP Trunk) on this SIP Endpoint ?

try below:

voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
supplementary-service h450.12
sip
  bind control source-interface FastEthernet0/0 (or 172.xx.xx.xx  interface IP)
  bind media source-interface FastEthernet0/0
  registrar server expires max 3600 min 120

if still issue, enable below debugs and attach here (capture for both incoming and outgoing):

debug voip ccapi inout
debug ccsip messages

HTH

Arun

[Pratik Jhaveri]

Following should hopefully do the job:

allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip

Binding the interface wouldn't be a very good idea as it not only sources the SIP source address but also forces to listen for SIP invites only on that interface. If your SIP trunk is terminating on a different interface than the binded interface, it will drop the SIP packets

If the above does not work, then go with debugs that Arun has suggested.

HTH

Regards,

Pratik Jhaveri