We have a CUCM ver 22.214.171.12400-9.
For our PSTN access we have the following scheme:
CUCM <========> Cisco2911 <======> ISP-------> PSTN
H323 tk SIP tk
Our Cisco2911 is running ver 150-1.M4.
We configure Cisco 2911 as H323 gateway on CUCM, and we established a SIP tk with our ISP.
When we tested incoming calls we received them at router but they are unable to acceess CUCM,
When we tested outgoing calls we got reorder tone and they did not reach our voice gateway.
We run "debug cch323 all" and tested with one incoming and one outgoing call, and we got the out on attached file "debug h323 forum"
CUCM Ip : 10.1.2.2
Router's H323 int: 10.2.2.1
Do you have any idea about the root cause?
Can be an access list problem preventing proper H323 traffic??
Thanks in advanced
seems to be problem with access list.
Reason code says; Indicates that the called party cannot be reached because the network that the call has been routed through does not serve the desired destination.
Thanks Suresh and Chintan !!
Today I completed some changes on:
a) router: change dial-peer configuration, changed CODEC towards CUCM.
b)CUCM: certify proper configuration/registration of gateway, MTP, Transconder, and MRG.
And I got the following results:
1) Inbound calls: destination Iphone keeps ringing, no matter amount of times I answer the call. The ringing stops when I finish the call on external Phone.
2) Outbounf calls: the external Phone receives the call and rings, once I take the call It drops.
It seems to be a CODEC problem, but I have tested with ISP getting the same bad behavior.
I run at the same time the followin debugs:
debug ccsip message
debug ccsip error
debug cch323 all.
I am attaching the file "debug h323 sip Forum.txt" with the output of an inbound call from PSTN 7872067379 to my pilot 7877981849, and the file "partial config.txt" with Voice related configuration
I hope this can help to detect the cause of this problem !!!!
Thanks in advanced
The most common reason why I've seen one side hearing ringback despite the other side answering is the "Wait for Far End TCS" being checked on the H323 gateway page on CCMAdmin. Uncheck this and it should work fine. When it's checked, CUCM waits for the gateway to send its TCS before sending the gateway its own TCS.
This checkbox was only meant for certain use cases, so uncheck it, and the incoming call should work.
Also, see if the "Inbound Fast Start" checkbox has been checked. Uncheck that.
I don't see the debugs for an outbound call in the attachments above.
For outbound, please take
debug voip ccapi inout
debug h225 asn1
debug h245 asn1
debg ccsip message
Another thing that I wanted to ask was, is there a reason why you are using H323 on the CUCM - GW side? Why not use SIP there as well so the 2911 does SIP -- SIP? This would be easier to troubleshoot and supports more features.
Thanks for your reply
I took your recommendation and established a SIP Tk between CUCM and Gw, so my new scheme is:
CUCM <== SIP tk ==> Cisco 2911<== Sip tk ==> ISP
But now, the problem changed:
a) Incoming calls: internal Ipphone rings, we take the call, got silence and after 10 secs the call drops.
b) Outgoing calls: Ip phone does not ring, when I take the call got silence on both sides., and althought y hang on the external phone, the Ip phone keeps waiting for the 1st ring until it drops the call.
When I run "debug ccsip message" and "debug ccsip error" I see :
1) When i run "show call active voice compact" the leg Gw-CUCM never is establihed.(only Gw-ISP leg)
2) On outgoing calls the CUCM keeps sending "INVITES" to CUBE router, no matter it receives "Session Progress 183" and "200 OK" messages from CUBE router (on behalf of PSTN peer).
I am attaching a pair of files with debugging output.
Any help is welcome.
I didn't find requested logs on your previous post , so i came back to this duplicate post.
I just looked at sip logs and its look like configuration is messing up your calls. Can you please attach router config and also debug voice ccapi inout for both the calls.
As you moved from H323 to SIP but you didn't change your dial-peer protocol and it is still working as h323.
Put session protocol sipv2 under dial-peer 999, 2 and 9002 and remove voice class voice-class h323 1 and outbound-proxy ipv4:172.29.1.10.
Collect the logs once again for inbound and outbound.
Manish thanks again.
I made recommended changes but still get the same behavior, and when I try with incoming call from PSTN I got the error " the number you have called is unavailable", once I remove the command "session protocol sipv2" on dial-peer 9002 (pointing to CUCM) the call can arrive to internal IpPhone but with the initial problem I got silence for 10 seconds until the call is dropped .
Here you have the debug taken.
I have tested tcp 5060 from router to CUCM, but I am not sure if UDP 5060 is open.
Do you know how can I verify if my CUCM is receiving properly the SIP/H323 messages from my CUBE???
I know how download RTMT reports , but I am not sure how read them.