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

Nicholas Samios nsamios at staff.iinet.net.au
Tue Jan 24 17:46:29 EST 2012


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
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <https://puck.nether.net/pipermail/cisco-voip/attachments/20120124/b2c0fd22/attachment.html>


More information about the cisco-voip mailing list