01-15-2014 04:57 AM - edited 03-16-2019 09:15 PM
hello all
actually i have a very confusing issue on connecting CME and CUCM.
i have CUCM ver 8.6.2 and cme ver 8.6 (on 2821), i try to connect this two call processing server over WAN Link with H323 protocol,
so i add CME as H323 Gateway and config my CME as below,
the problem is there is only one way voice connection or even second party answer the phone , first one give the fast busy tone
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
no supplementary-service h450.7
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
h323
interface GigabitEthernet0/0
ip address 192.168.100.12 255.255.255.0
duplex auto
speed auto
h323-gateway voip interface
h323-gateway voip bind srcaddr 192.168.100.12
dial-peer voice 13 voip
description HQ Outgoing
destination-pattern [1-4]..
session target ipv4:192.168.1.200
codec g711ulaw
Thanks in Advance
Solved! Go to Solution.
01-22-2014 12:45 AM
Hi John,
i call from CME to CUCM phone rings but even i asnwer the call it going to fast busy - This is codec issue , codec negotiation happens after call answers and thats the reason why call getting disconnected.
Can you check region under device pool which is assigned to the trunk and device pool/region of CUCM ip phone. What codec is used between them ?
And also can you attach your CME router config.
Thanks
Manish
01-15-2014 05:02 AM
Make sure your IP routing is configured properly as one way audio is 99% of the time related to network routing issues.
Chris
Sent from Cisco Technical Support iPhone App
01-15-2014 05:11 AM
Hi John,
You can try adding "voice rtp send-recv" , if you IOS version is 12.2T or earlier. But as Chris pointed one-way audio always been a routing issue.
Thanks
Manish
01-15-2014 05:40 AM
Dear Manish
i use
c2800nm-adventerprisek9-mz.151-4.M1.bin
01-15-2014 05:40 AM
actually i try th ping and trace from both CME and CUCM and its successful
i also try the SIP Trunk, it better from h323 and at this time if i call from CME to CUCM i have 2way call and if i call from CUCM to CME, CUCM caller can not hear my vocie over CME, but i can hear what he said
and one thing remain is i set the SIP Profile in CUCM on both CIsco and Standard
below is my config in SIP Trunk
voice class h323 1
h225 timeout tcp establish 3
h225 timeout setup 3
dial-peer voice 33 voip
description *** Outoing call to CUCM ***
destination-pattern [1-4]..
session protocol sipv2
session target ipv4:192.168.1.200
session transport udp
voice-class h323 1
dtmf-relay sip-notify
no vad
!
dial-peer voice 34 voip
description *** Incoming call from CUCM ***
session protocol sipv2
incoming called-number .
voice-class h323 1
dtmf-relay h245-alphanumeric
no vad
!
!
sip-ua
sip-server ipv4:192.168.1.200
01-15-2014 06:05 AM
Hi John,
RTP always exchange between end points (if any extra feature/service is not included) . So I would check IP phone gateway from each site and check their connectivity with each other and IP phones.
Thanks
Manish
01-21-2014 01:17 AM
Dear Manish
as you said i check the connectivity and result is the same and i have ping all over the netwrok
so i make a change to the topology as below
ip-phone <---> CUCM8.6<----> "GRE TUNNEL" <---> CUCME8.6 <----> ip-phone
i change the trunk to Intracluster trunk(non-gatekeeper) and at this situation i can call from CUCM to CUCME but if i call from CUCME to CUCM phone rings but if i answer the phone it goes to fast busy and disconnected
below is my CUCME configuraion:
interface GigabitEthernet0/0
ip address 10.12.12.12 255.255.255.0
duplex auto
speed auto
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
no supplementary-service h450.7
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
h323
dial-peer voice 14 voip
destination-pattern 4..
session target ipv4:10.10.10.100
no vad
THANKS in advance
John
01-21-2014 04:52 AM
Hi John,
Please add this under GigabitEthernet0/0 (if its an uplink interface towards CUCM)
h323-gateway bind srcaddr x.x.x.x
Where x.x.x.x same IP address defined when you add a gateway to a CUCM.
And also do you have an incoming dial-peer configured on CME for the call coming from CUCM ?
Thanks
Manish
01-21-2014 11:17 PM
Dear Manish
because i use BACD on CME for IVR, even i use h323-gateway bind srcaddr incomming calls ddint go to IVR and disconnected, i test it before, and for incomming dial peer, i think it can use default dial-peer for CUCM, but i will test it and give you the report
John
01-21-2014 11:56 PM
i also think maybe the RTP session from CME to CUCM drop through the WAN after call establish
because if i call from CME to CUCM phone rings but even i asnwer the call it going to fast busy
and also there is another issue, i registre 1 phone on CUCM but in other side and in my LAN (so there is no TUNNEL if they want to call each other, as you know RTP session establish between two phone after the call), but its the same and i get the fast buy after hangup the phone
01-22-2014 12:45 AM
Hi John,
i call from CME to CUCM phone rings but even i asnwer the call it going to fast busy - This is codec issue , codec negotiation happens after call answers and thats the reason why call getting disconnected.
Can you check region under device pool which is assigned to the trunk and device pool/region of CUCM ip phone. What codec is used between them ?
And also can you attach your CME router config.
Thanks
Manish
01-27-2014 01:50 AM
Thank you Dear Manish
i config the Codec Class and assign it to dial-peers,, so its going to be OK
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