This is my first time posting so please forgive any errors. We are experiencing one way audio issues in our newly implemented SIP platform. I came across this thread and think I may be on the right track as to what is going on. I am very new to Cisco Call Manager as most of my experience has been in the Avaya world. Currently, our organization has multiple phone systems and we will be consolidating shortly. For the sake of this comment, I will talk of our Cisco and Avaya as this is where we are experiencing problems.
On the west coast, we have a Cisco Call Manager and on the East Coast, Avaya. We have AT&T SIP trunks in both locations. On the west coast is where I think the problem lies. Our connection is as follows:
Call Manager -> CUBE -> ATT SIP RTR (hope that makes sense)
The problem that we are facing is that west coast users can hear east coast but, east cannot hear west. This happens whether the call is inbound or outbound.
here is my config... are we missing something or is there a misconfiguration that you may see?
interface GigabitEthernet0/0/0
description *** Voice Gateway LAN ***
ip address x.x.x.x 255.255.255.0
negotiation auto
!
interface GigabitEthernet0/0/1
description TO AT&T_Voice ROUTER
ip address x.x.x.x 255.255.255.248
negotiation auto
!
!
router eigrp 100
network x.x.x.x 0.0.0.255
!
ip forward-protocol nd
no ip http server
no ip http secure-server
ip tftp source-interface GigabitEthernet0
ip route x.x.x.x 255.255.0.0 y.y.y.y
ip ssh version 2
!
!
!
!
!
!
control-plane
!
!
!
!
!
!
mgcp behavior rsip-range tgcp-only
mgcp behavior comedia-role none
mgcp behavior comedia-check-media-src disable
mgcp behavior comedia-sdp-force disable
!
mgcp profile default
!
sccp local GigabitEthernet0/0/0
sccp ccm x.x.x.x identifier 1 priority 1 version 7.0
sccp ccm x.x.x.x identifier 2 priority 2 version 7.0
sccp
!
sccp ccm group 1
associate ccm 1 priority 1
associate ccm 2 priority 2
associate profile 1 register MTP_CUBE_SCDC
associate profile 2 register CFB_CUBE_SCDC
associate profile 3 register XCODE_CUBE_SCDC
keepalive retries 5
switchover method immediate
switchback method immediate
switchback interval 15
!
!
!
dspfarm profile 3 transcode
codec g729abr8
codec g729ar8
codec g711alaw
codec g711ulaw
maximum sessions 15
associate application SCCP
!
dspfarm profile 2 conference
codec g729br8
codec g729r8
codec g729abr8
codec g729ar8
codec g711alaw
codec g711ulaw
maximum sessions 15
associate application SCCP
!
dspfarm profile 1 mtp
codec g711ulaw
codec pass-through
maximum sessions software 150
associate application SCCP
!
dial-peer voice 100 voip
description INBOUND From-CUCM
session protocol sipv2
incoming called-number 7T
voice-class codec 1
voice-class sip bind control source-interface GigabitEthernet0/0/0
voice-class sip bind media source-interface GigabitEthernet0/0/0
dtmf-relay rtp-nte sip-kpml sip-notify
no vad
!
dial-peer voice 101 voip
session protocol sipv2
incoming called-number [2-9]..[2-9]......$
voice-class codec 1
voice-class sip bind control source-interface GigabitEthernet0/0/1
voice-class sip bind media source-interface GigabitEthernet0/0/1
dtmf-relay rtp-nte
no vad
!
dial-peer voice 200 voip
description OUTBOUND to-CUCM-SUB
preference 1
destination-pattern [2-9]..[2-9]......$
session protocol sipv2
session target ipv4:x.x.x.x.
voice-class codec 1
voice-class sip bind control source-interface GigabitEthernet0/0/0
voice-class sip bind media source-interface GigabitEthernet0/0/0
dtmf-relay rtp-nte
no vad
!
dial-peer voice 201 voip
description OUTBOUND to-CUCM-PUB
preference 2
destination-pattern [2-9]..[2-9]......$
session protocol sipv2
session target ipv4:x.x.x.x
voice-class codec 1
voice-class sip bind control source-interface GigabitEthernet0/0/0
voice-class sip bind media source-interface GigabitEthernet0/0/0
dtmf-relay rtp-nte
no vad
!
dial-peer voice 300 voip
description OUTBOUND to-ATT
translation-profile outgoing OUTGOING-WAN
preference 1
destination-pattern 7T
session protocol sipv2
session server-group 111
voice-class codec 1
voice-class sip bind control source-interface GigabitEthernet0/0/1
voice-class sip bind media source-interface GigabitEthernet0/0/1
dtmf-relay rtp-nte
no vad
!
dial-peer voice 301 voip
description INBOUND from-ATT
preference 2
session protocol sipv2
incoming called-number 650.......
voice-class codec 1
voice-class sip bind media source-interface GigabitEthernet0/0/1
dtmf-relay rtp-nte
no vad
!
!
sip-ua
retry invite 2
retry bye 2
retry cancel 2
retry register 10
