[cisco-voip] CUBE not requesting codec... call fails... need to force SDP in invite...

[Nick Matthews]

Cool, glad it's working.  I've heard a ton of crazy things about CUBE,
think this is the first that it doesn't do SIP-SIP.

Do you have this command:
voice service voip
 sip
  allow connection sip to sip

SIP to SIP is preferred because it's smoother and that's where all the
development is happening so the new features are all SIP-SIP.

-nick

On Tue, Jan 24, 2012 at 9:54 PM, Jonathan Charles <jonvoip at gmail.com> wrote:
> Because, as far as I can tell, Cisco does not support SIP to SIP on the
> CUBE... and it doesn't work.
>
> You need to be H.323 to the CUBE, then SIP to the provider.
>
>
> On Tue, Jan 24, 2012 at 4:46 PM, Nicholas Samios
> <nsamios at staff.iinet.net.au> wrote:
>>
>> Why not make the CUCM <-> CUBE a SIP Trunk instead of H323?
>>
>>
>>
>> Could try below on the CUBE LD peer which should force the DO to EO to the
>> ITSP;
>>
>>
>>
>> voice-class sip early-offer forced
>>
>>
>>
>> From: cisco-voip-bounces at puck.nether.net
>> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Jonathan Charles
>> Sent: Wednesday, January 25, 2012 9:03 AM
>> To: cisco-voip at puck.nether.net
>> Subject: [cisco-voip] CUBE not requesting codec... call fails... need to
>> force SDP in invite...
>>
>>
>>
>> From the attached debug, we are sending an invite without codec info;
>> carrier comes back and says G729 and we drop it. Users hear one ring and
>> fast busy.
>>
>>
>>
>> Route is running SP Services 15.1.4M1
>>
>>
>>
>> Network is Phone to CCM, H.323 to CUBE, SIP to Provider.
>>
>>
>>
>> LD calls work cuz they are G.711 all the way, 800 numbers fail (cuz the
>> carrier points it at a G.729 trunk).
>>
>>
>>
>>
>>
>> I have implemented the fix here:
>>
>>
>>
>> http://www.markholloway.com/blog/?p=1325
>>
>>
>>
>> But it doesn't work.
>>
>>
>>
>> The H.323 gateway does NOT have MTP required checked currently (it made no
>> difference.
>>
>>
>>
>>
>>
>>
>>
>>
>>
>> [relevant config]
>>
>> !
>>
>> voice service voip
>>
>>  ip address trusted list
>>
>>   ipv4 10.0.0.0 255.0.0.0
>>
>>   ipv4 172.16.0.0 255.240.0.0
>>
>>   ipv4 172.20.0.0 255.255.255.0
>>
>>  allow-connections h323 to h323
>>
>>  allow-connections h323 to sip
>>
>>  allow-connections sip to h323
>>
>>  allow-connections sip to sip
>>
>>  fax protocol pass-through g711ulaw
>>
>>  h323
>>
>>   h225 display-ie ccm-compatible
>>
>>  modem passthrough nse payload-type 101 codec g711ulaw
>>
>>  sip
>>
>>   bind control source-interface GigabitEthernet0/0
>>
>>   bind media source-interface GigabitEthernet0/0
>>
>> !
>>
>> voice class codec 1
>>
>>  codec preference 1 g711ulaw
>>
>>  codec preference 2 g729r8
>>
>> !
>>
>> dial-peer voice 201 voip
>>
>>  translation-profile outgoing Last10
>>
>>  preference 1
>>
>>  destination-pattern ^1[1-9].........
>>
>>  session protocol sipv2
>>
>>  session target ipv4:72.11.192.82
>>
>>  session transport udp
>>
>>  voice-class codec 1
>>
>>  no voice-class sip pass-thru content sdp
>>
>>  dtmf-relay rtp-nte
>>
>>  no vad
>>
>> !
>>
>> dial-peer voice 202 voip
>>
>>  translation-profile outgoing Last10
>>
>>  preference 2
>>
>>  destination-pattern ^1[1-9].........
>>
>>  session protocol sipv2
>>
>>  session target ipv4:72.11.193.82
>>
>>  session transport udp
>>
>>  no voice-class sip g729 annexb-all
>>
>>  no voice-class sip pass-thru content sdp
>>
>>  dtmf-relay rtp-nte
>>
>>  no vad
>>
>> !
>>
>>
>>
>>
>>
>> So, the question is, how do I force an SDP specifying G.711 in the invite?
>>
>>
>>
>>
>>
>>
>>
>>
>>
>> Jonathan
>
>
>
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