I have a SIP DID from Vitelity connected to my 2821 H323 gateway and CUCM 7.1.
When I make a call over SIP I don't hear any ringback, but the call is routed correctly and the call works fine once the called-party answers.
Here is some of my gateway config;
voice service voip
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
fax protocol cisco
voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g729r8
voice class h323 1
h225 timeout tcp establish 3
description ******H323 GATEWAY ADDRESS******
ip address 10.10.10.1 255.255.255.255
h323-gateway voip interface
h323-gateway voip h323-id C2821
h323-gateway voip bind srcaddr 10.10.10.1
dial-peer voice 1 voip
voice-class codec 1
session target ipv4:192.168.1.10
incoming called-number .
dial-peer voice 2 voip
progress_ind setup enable 3
progress_ind alert enable 8
voice-class codec 1
session protocol sipv2
session target sip-server
session transport udp
incoming called-number 9T
timer receive-rtp 1200
credentials username [removed] password 7 [removed] realm asterisk
authentication username [removed] password 7 [removed] realm asterisk
retry invite 2
retry register 2
registrar ipv4:126.96.36.199 expires 3600
Any ideas are appreciated, thanks in advance!
Hi , i got the issue resolved at one of our customer site.. we had to configure an IOS MTP and also restarted the IP Voice media streaming service
You should make sure that the codec you have chosen is being negotiated ( and available on the MTP). If you're trying to use G.729 you'll need a software MTP on a router.
Also make sure that the MRGL has access to the MTP. The INVITE that is sent to the PBX should have information under it, an SDP. It should include IP addresses, codecs, etc.
This isn't very easy to explain -
Your H323 side is not doing fast start and you're getting media in your 1xx message to the Asterisks.
You need to do media negotiation in order to hear ringback in this case. The gateway cannot send its media to the Asterisks in this case because it cannot send any type of response at this point in the call flow. If the Asterisks supported rel1xx messaging, you would be able to.
The only workaround is to check 'MTP Required' on the H323 gateway in CUCM and check 'Enable Outbound Faststart'.
If you're determined not to use MTPs, you will need to switch your trunk to CUCM to SIP. In 12.4(20)T and later you can use the CUBE feature 'DO-EO', which is the 'early-offer forced' command.