01-14-2014 04:52 AM - edited 03-16-2019 09:14 PM
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
Solved! Go to Solution.
01-14-2014 05:18 AM
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?
Please rate all useful posts
"The essence of christianity is not the enthronement but the obliteration of self --William Barclay"
01-14-2014 05:25 AM
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
Please rate all useful posts
"The essence of christianity is not the enthronement but the obliteration of self --William Barclay"
01-14-2014 04:56 AM
01-14-2014 05:18 AM
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?
Please rate all useful posts
"The essence of christianity is not the enthronement but the obliteration of self --William Barclay"
01-14-2014 05:22 AM
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
01-14-2014 05:25 AM
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
Please rate all useful posts
"The essence of christianity is not the enthronement but the obliteration of self --William Barclay"
01-14-2014 05:29 AM
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
01-14-2014 09:48 AM
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.
Discover and save your favorite ideas. Come back to expert answers, step-by-step guides, recent topics, and more.
New here? Get started with these tips. How to use Community New member guide