Hi guys,
I am looking for some advice regarding a proof of concept setup we are test lab-ing before we commit to getting a CUBE.
Please see attachment for our test lab setup..
We have created a SIP trunk from our CUCM7.1.5 to another PBX ( Asterisk 2.6 ) with a Cisco 2851 inbetween ( running Version 15.1(1)T4 ) acting as a CUBE..
Calls can be successfully made across this SIP trunk and we get two way audio..
One of the key features of the CUBE which we are interested in is the ability for us to insert our own Music on Hold feed.. We currently successfully do this by applying an ACL blocking the multicasting stream coming from CUCM and then reinserting our own stream further down the line.
So, going back to our SIP call.. We can make a call from CUCM across to our test PBX ( Asterisk ).
When we place the call on hold.. We can see our test CUBE pulling the correct multicast stream and the packets in/out increment..
TestCUBE#sh ccm-manager music-on-hold
Current active multicast sessions : 1
Multicast RTP port Packets Call Codec Incoming
Address number in/out id Interface
===================================================================
239.1.10.9 20480 792/792 21185 g711ulaw Gi0/0
However, RTP does not send any further and is not received by the PBX test phone.
If I run a packet capture we can see that the phone does receive the INVITE at the point of placing the call on hold.
Has anyone got a similar set up and able to advise what I may be missing here please.
Ive included parts of our CUBE config below
we are using u law codec.. But experienced the same issue with a law
We've set the sip trunk to be in pass-through mode.. Have tested in flow around too, and this did not work
As only test environment, we only created a single dial peer to route 0132 calls across SIP trunk
Cisco 2851 setup
Version 15.1(1)T4
voice call send-alert
voice rtp send-recv
!
voice service voip
allow-connections sip to sip
no supplementary-service sip refer
supplementary-service media-renegotiate
redirect ip2ip
sip
rel1xx disable
min-se 360
header-passing
subscription maximum originate 2
pass-thru headers unsupp
pass-thru content unsupp
!
voice class media 1
media transcoder high-density
media forking
!
voice class codec 1
codec preference 1 g711ulaw
!
interface GigabitEthernet0/0
ip address 10.12.222.240 255.255.255.0
no ip redirects
no ip proxy-arp
ip pim query-interval 1
ip pim sparse-mode
!
ccm-manager music-on-hold bind GigabitEthernet0/0
!
dial-peer voice 10 voip
description **Outgoing to Asterisk extn 0132**
destination-pattern 0132
session protocol sipv2
session target ipv4:10.15.1.64
session transport udp
voice-class codec 1
dtmf-relay rtp-nte
ip qos dscp cs3 signaling
no vad
!
!
dial-peer voice 1000 voip
description **Inbound from Asterisk **
session protocol sipv2
session target ipv4:10.15.1.64
session transport udp
incoming called-number 8506
voice-class codec 1
dtmf-relay rtp-nte
ip qos dscp cs3 signaling
no vad
!
dial-peer voice 1001 voip
description *** Asterisk -> CUCM_PUB ***
preference 10
destination-pattern 8506
session protocol sipv2
session target ipv4:10.15.223.10
session transport tcp
voice-class codec 1
dtmf-relay rtp-nte
ip qos dscp cs3 signaling
clid strip name
!
!
sip-ua
sip-server ipv4:10.15.1.64
Call Manager 7.1.5 Config.
Routing pattern created for dialled 0132, pointing to SIP Trunk
SIPTrunk created in CUCM routing this call to CUBE, 10.12.222.240
MTP has been disabled on the SIP trunk
U law.
