CME-- Call manager SIP call is not working

[renji joy]

Hello Team,,

We are facing the issue with Cisco CME with call manager .

If we are trying the SIP trunk calls  from call manager not going to cme ,from cme to call manager sip call is connecting but there is no audio.

with the same setup H.323 calls going on both the way without any issue.

Configuration Done.

SIP trunk and route pattern configured and pointed to the CME ip addres..

Dial-peer has been configured in the cme router.

Any thing is missing and suggestion please...

[Nadeem Ahmed]

Do you have CUCM trace for calls not connecting from CUCM ----SIP TRUNK----CME and debugs logs CME------SIP TRUNK----CUCM.

CUCM to CME how they are connected? are they over WAN, VPN or LAN?

Can you please show run from CME? also check if you have applied the bind command on voice service voip or dial-peer level pointing to CUCM.

Br,

Nadeem Ahmed

[renji joy]

Hello ,

Thanks for the update 

please find the attached sh runn for the cme.. 

[Nadeem Ahmed]

Are you using these dial-peer for calling from CUCM to CME and vice-versa. In CUCM Cluster how many nodes are there? secondly can you make a test from both side and collect debugs and CUCM TRACES.

for Gateway Debugs run below debugs

1) debug ccsip messages

2) debug voice ccapi inout

from CUCM

enable detailed traces

dial-peer voice 3 voip
 description TEST
 preference 2
 destination-pattern 007........
 session target ipv4:172.16.5.100
 incoming called-number .
 dtmf-relay rtp-nte
 codec g711ulaw
 no vad
!
dial-peer voice 2 voip
 corlist outgoing localcalls
 description "Natoinal Mobile "
 preference 2
 destination-pattern 007........
 session target ipv4:172.16.5.100
 codec g711ulaw
!
dial-peer voice 10 voip
 description "Moe HQ Dial-peer"
 preference 6
 destination-pattern 8...
 session target ipv4:172.16.5.100
 session transport udp
 incoming called-number .
 voice-class codec 1  
 dtmf-relay rtp-nte
 no vad
!

[renji joy]

Hello,

CUCM to CME connected over the VPN ,also tried to without VPN.

yes we are using the same dial-peer for calling.and there is two nodes in the cluster.

please note with the same setup H.323 calls going on both the way without any issue..

i will be sending the detailed logs for the reference.

[Pawan Srivastava]

To use SIP trunk between CME and CUCM, below are the recommendations:

1. From debug, CME received single digit from Phone and it looks like related to bug: CSCum05299.

Add below configuration and hard reset the phone:

                voice service voip

                no ip address trusted authenticate

2. Modify interface config:

#interface GigabitEthernet0/0.103

#no h323-gateway voip interface

#no h323-gateway voip bind srcaddr 172.18.3.129

3. In CUCM, SIP trunk destination IP address: 172.18.3.129 (GW voice vlan interface ip address)

4. In CUCM, make sure IP Phone and SIP trunk have same Device pool.

5. Modify dial-peer

#no dial-peer voice 10 voip

#dial-peer voice 10 voip

#description "Moe HQ Dial-peer"

#preference 6

#destination-pattern 8...

#session protocol sipv2

#session target ipv4:172.16.5.100

#incoming called-number .

#codec g711u

#no vad

Please test from SIP/SCCP phones, incoming/ outgoing calls and if any issue then share below information:

++ calling, called party number

++ enable 'debug voip ccapi inout', 'debug ccsip message', 'debug ccsip error'

PS: To avoid port/routing issue, test without VPN

[Carlo Poggiarelli]

Hi.

On your dialpeers you have to specify "protocol sipv2 " if you want to send or receive sip calls from/to CME

Please add it and let us know 

Thanks

Regards

Carlo

[Mohammed al Baqari]

Hi,

As mentioned by Carlo, you current dialpeers are using H323. To use SIP you need to apply the command 'session protocol sipv2' on each dialpeer. However, I don't think this is the only problem since this command is used for calls from CME. Calls to CME won't need this command.

Can you please do the following.

1. Start a call from CME to CUCM.

2. Share the output of 'debug ccsip messages' from the CME

3. Share the output of 'show voip rtp conn' from the CME

4. Share the output of 'show call active voice brief' from CME

5. Share the output of 'show sccp conn' from CME.

6. End the first call and try a call from CUCM to CME. Share the output of 'debug sip messages'

[renji joy]

Hello Carlo,

if applying the session protocol sipv2 command call is not going from CME.

Also if selecting SIP trunk in the route pattern call is not going from call manger

sharing the outputs also..

[renji joy]

Hello,

We have registered one branch ip phone directly to the HO call manager for testing ,but the same issue call is not going from call manager /and from the branch office call is connecting but there is no audio

[Mohammed al Baqari]

Please share the outputs which I requested to be able to assist.

[renji joy]

Hello,

As we mentioned earlier if applying the session protocol sipv2 command call is not going from CME.

Please find the

debg ccapi in out

debug ccsip messages

[Tagir Temirgaliyev]

u need inbound sip dial-peer too

[renji joy]

Can you please give the example of sip dial peer

Thanks

[renji joy]

Configured the in bound dial-peer,can you please verify

dial-peer voice 4 voip
 description incoming
 incoming called-number .
 codec g711ulaw
 no vad