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CUCM and CME Trunk connectivity

John Mayer
Level 1
Level 1

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

1 Accepted Solution

Accepted Solutions

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

View solution in original post

11 Replies 11

Chris Deren
Hall of Fame
Hall of Fame

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

Manish Prasad
Level 5
Level 5

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

Dear Manish

i use

c2800nm-adventerprisek9-mz.151-4.M1.bin

John Mayer
Level 1
Level 1

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

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

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

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

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

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

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

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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