Sip Trunk CUCM /ASA/ ITSP

[Leonardo Santana]

Hello

I have an sip trunk with an ITSP in CUCM, thats is the flow:

IP Phone -> CUCM -> SIP TRUNK  -> ASA -> ITSP

The outbound call is established but has no audio.

On ASA has a NAT from the outside interface do CUCM IP address

When the audio is established, is from CUCM to the SIP Proxy?

Inbound calls are not established

SIP/2.0 503 Service Unavailable

Via: SIP/2.0/UDP 66.33.147.146:5060;branch=z9hG4bK_1128038047_8107_1

From: <sip:000121263600@66.33.147.146:5060>;tag=1128038047_C

To: <sip:2766@192.168.10.210:5060>;tag=1599180641

Date: Mon, 13 Jan 2014 19:46:11 GMT

Call-ID: sbcsipuac.2_206.20.7.11_b12sb09_1_1_2014011314433251_1128038047_218711

CSeq: 1001 INVITE

Allow-Events: presence

Warning: 399 SRVRCUCMPUB "Unable to find a device handler for the request received on port 5060 from 66.33.147.146"

Content-Length: 0

Regards

Leonardo Santana

[Hermanus Janse van Vuuren]

Have a look at the following Thread

https://supportforums.cisco.com/thread/276600

[Ayodeji Okanlawon · Accepted solution]

This error "Warning: 399 SRVRCUCMPUB "Unable to find a device handler for the request received on port 5060 from 66.33.147.146" suggests that CUCM is receiving traffic from an ip address that is not configured on the sip trunk facing the ITSP..

What is the ip address that you have specified on the sip trunk?

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"The essence of christianity is not the enthronement but the obliteration of self --William Barclay"

[Leonardo Santana]

Hello

For the inbound calls a found the issue, the ITSP have differents IPs Address to receive and make the call.

I created an specif SIP Trunk for inbound and one for outbound, the calls are now established but both (inbound and outbound) with no audio.

Regards

Leonardo Santana

[Ayodeji Okanlawon · Accepted solution]

I suggest you look at the firewall here..Check that you have ip routing to the ip address where media is terminated..Check your ASA logs, something is not right there

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"The essence of christianity is not the enthronement but the obliteration of self --William Barclay"

[Leonardo Santana]

Thanks

I will check that, IP Routing is enabled from the voice network i can reach the SIP Proxy Address, but i think the RTP is blocked, i ask the Security Team to check this.

Thanks again.

Regards

Leonardo Santana

[kelvin.blair]

Leanardo suggested IP Routing which is one thing to check.  The next thing to check is the sip inspection rules. Doing SIP Inpection on an ASA for this type of design can cause you issues..  So you might want to turn that off in the ASA as well.