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...

[renji joy]

We can see in the same logs correct info also..

please check...

INVITE sip:8969@172.16.5.100:5060 SIP/2.0

Via: SIP/2.0/UDP 172.18.3.129:5060;branch=z9hG4bK11BC6

Remote-Party-ID: "5002" <sip:5002@172.18.3.129>;party=calling;screen=yes;privacy=off

From: "5002" <sip:5002@172.18.3.129>;tag=34507C-710

To: <sip:8969@172.16.5.100>

[Tagir Temirgaliyev]

do u have only one cucm server? where are publisher? where are subscribers?

[renji joy]

We have two ucm servers one publisher and subscriber both are in same locations.

Thanks

[Tagir Temirgaliyev]

subscriber   is 172.16.5.100  ?

deb ccsip mess

and make call from 5002 to 5000

and make call from 5002 to 3002

[renji joy]

172.16.5.100 is the publisher ,both are the TFTP servers,

sorry the site is down now.

Internally all the calls working normally.

i will take the debug and post you...

Thanks

[Pawan Srivastava]

From shared ccsip debugs and new_rtmt output and it appear as network issue.
++ CUCM received the call Invite from Gateway:
2015/08/01 15:00:00.222|SIPT|0|UDP|IN|172.16.5.100|5060|Nangaharsiptrunk|172.18.3.129|51394|1,100,220,1.176^172.18.3.129^*|9873726|CFA4013E-36A511E5-80C1F753-AFCC55FB@172.18.3.129|INVITE

++ CUCM responded with Trying (100)
2015/08/01 15:00:00.223|SIPT|0|UDP|OUT|172.16.5.100|5060|Nangaharsiptrunk|172.18.3.129|5060|1,100,220,1.176^172.18.3.129^*|9873727|CFA4013E-36A511E5-80C1F753-AFCC55FB@172.18.3.129|100 Trying

++ Later CUCM responded with 180, 183, 200 Messages but nothing reached to Gateway.
2015/08/01 15:00:00.294|SIPT|20725455|UDP|OUT|172.16.5.100|5060|Nangaharsiptrunk|172.18.3.129|5060|1,100,57,1.3403655^172.16.75.236^*|9873731|CFA4013E-36A511E5-80C1F753-AFCC55FB@172.18.3.129|180 Ringing
2015/08/01 15:00:05.862|SIPT|20725455|UDP|OUT|172.16.5.100|5060|Nangaharsiptrunk|172.18.3.129|5060|1,100,62,29507.1^*^*|9873734|CFA4013E-36A511E5-80C1F753-AFCC55FB@172.18.3.129|183 Session Progress

++ At last call was cancelled by calling SIP phone then gateway cancelled the call with CUCM:
2015/08/01 15:00:13.913|SIPT|20725455|UDP|IN|172.16.5.100|5060|Nangaharsiptrunk|172.18.3.129|51394|1,100,220,1.177^172.18.3.129^*|9873755|CFA4013E-36A511E5-80C1F753-AFCC55FB@172.18.3.129|CANCEL

++ Please check the network why SIP messages (180,183,200) blocked or dropped from CUCM to Gateway. (One possibility is mentioned SIP messages with SDP and blocked by FW).

[renji joy]

Thanks for your support,

As you aware we have tested the same setup in our internal network with firewall ,and its work normally ,but when is connecting with service provider media( Data E1)

and facing this issue.

Thanks

[Pawan Srivastava]

With current logs, it is not voice configuration issue and more over data configuration issue.

My understanding about Internal outbound call flow as follows:

SIP Phone --> CME --> SIP --> CUCM --> 8969 (another Phone SEPE0D173E58BEC in CUCM).

Please clarify, do above outbound call flow works or not. If yes then which part of the voice component communication are we changing from Internal network to Service Provider Media (Data E1). Please share the complete call flow for non-working calls.

[renji joy]

Hello,

SIP Phone --> CME --> SIP --> CUCM --> 8969 (another Phone SEPE0D173E58BEC in CUCM)

​The mentioned call flow is correct.

Actually we have connected one another cme(same product and same version of cme,same configuration and same model ip phone ) in our internal network .

One physical connectivity is going from our ASR router to microwave IDU (for E1 link)

For testing  we are removing ASR router cable and directly connecting to the testing cme router. and configured sip trunk between the test cme.

that time we can able to make a call bi-directional without any issue.

When we are connecting ASR router to the IDU for the wan connectivity facing the issue.

Thanks

[renji joy]

please find the new cme show run

I have added the session protocol sipv2 in the dial-peer.but the same issue call is not going from the cme

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