SIP to H323 to SCCP Call are not connecting after IVR configured on Cisco Voice Gateway 3825

[ahmed-mujeeb]

Hi Everyone,

I have set a SIP trunk from a ITSP provider on Cisco 3825 router, It sas successfully registered with ITSP.

1) we can successfully made a call form internal network to PSTN.

2) But when trying to call from PSTN to internal network, IVR played but when we dialed the desired extension the ring continuous to go, hence the IP Phone rings but when anyone pick up there was no voice and that time the caller is continuously listening the ringing tone. IVR is ios based on Cisco router.

My call flow is as follows,

PSTN ==> SIP(ITSP) ==> H323(VG to CUCM) ==> SCCP(IP Phones)

Kindly Help me,

[ahmed-mujeeb]

i just experienced now that, in an inbound call on Sip PRI the issue is the call connects properly but after some time we are unable to hear any voice but the call remain connected.

[Ayodeji Okanlawon]

Remove the inbound fast start and outbound fast start on the gateway. Reset  the gateway. Use sip for the IVR calls as suggested. In the future consider using sip for all calls to cucm. 

[ahmed-mujeeb]

I tried establishing SIP trunk following the same configuration you shared but it doesn't established and call still uses H323 gateway for outgoing and incoming.

[Ayodeji Okanlawon]

The sip solution was just to test incoming call to 2625 not for outbound calls. 

Please do the sip configuration again and enable debug ccsipmessages 

Test again and send the logs

[ahmed-mujeeb]

Hi Ayodeji,

I have set the sip trunk again and disable the Inbound FastStart on H323 gateway, now again the same issue occurs as before the call doesn't connect.

called number = 2625

calling number = 34536571

[Ayodeji Okanlawon]

Ahmed,

The called number is not 2625 but 2633..So it will not use the sip trunk. The 2625 is for test purposes..So you need to make sure its 2625 that is dialled after the IVR plays..

Call comes into 00922137131444, which is your IVR. Then the user entered 2633, instead of 2625..

  Match Rule=DP_MATCH_DEST; Called Number=2633
Sep 17 11:53:15.855: //-1/7F6BBDB18C53/DPM/dpMatchPeersCore:
   Result=Success(0) after DP_MATCH_DEST
Sep 17 11:53:15.855: //-1/7F6BBDB18C53/DPM/dpMatchSafModulePlugin:
   dialstring=2633, saf_enabled=1, saf_dndb_lookup=1, dp_result=0
Sep 17 11:53:15.855: //-1/7F6BBDB18C53/DPM/dpMatchPeersMoreArg:
   Result=SUCCESS(0)
   List of Matched Outgoing Dial-peer(s):
     1: Dial-peer Tag=2001
     2: Dial-peer Tag=2002
     3: Dial-peer Tag=100

[ahmed-mujeeb]

Hi Ayodeji,

I am registering a CIPC on VPN on my premises and now going to test the call on it and share you the debugs.

[ahmed-mujeeb]

Hi, I have created this dial-peer for my CIPC

dial-peer voice 4199 voip
 destination-pattern 4199
 session protocol sipv2
 session target ipv4:172.20.11.10
 voice-class codec 1  
 dtmf-relay rtp-nte sip-kpml
 no vad

called number = 34311909

calling number = 4199

debugs are attached

[Ayodeji Okanlawon]

The call is using sip now but its failing because you are using outbound proxy, which means that this call is sent back toy our ITSP..

Please add the following to oyur dial-peer

dial-peer voice 4199 voip

no voice-class sip outbound-proxy

[ahmed-mujeeb]

Hi Ayodeji,

Ihave added the no voice-class sip outbound-proxy to the dial peer but still same issue presists, the call didn't connected. I am sharing logs.

dial-peer config

dial-peer voice 4199 voip
 destination-pattern 4199
 session protocol sipv2
 session target ipv4:172.20.11.10
 voice-class codec 1  
 no voice-class sip outbound-proxy   
 dtmf-relay rtp-nte sip-kpml
 no vad

[Ayodeji Okanlawon]

We are getting closer. Now the call goes to cucm but it's using the wrong interface on the Cube. It looks like you used the gig0/0 IP of CUBE ON CUCM. Add the ff commad. 

dial-p v 4199

voice-class sip bind control so gig0/0

voice-class sip bind media so gig0/0

[ahmed-mujeeb]

still facing the same issue. dial-peer config is as follows and debugs are attached

dial-peer voice 4199 voip
 destination-pattern 4199
 session protocol sipv2
 session target ipv4:172.20.11.10
 voice-class codec 1  
 no voice-class sip outbound-proxy   
 voice-class sip bind control source-interface GigabitEthernet0/0
 voice-class sip bind media source-interface GigabitEthernet0/0
 dtmf-relay rtp-nte sip-kpml
 no vad

[Ayodeji Okanlawon]

What IP address did you use on the sip trunk in cucm? 

Please use this. Change it to gig 0/0 IP and reset the trunk. Make sure you reset the trunk. Also ensure the inbound css can call the partition of 4199

172.20.11.50

[ahmed-mujeeb]

Ayodeji now sip trunk works, the IP address was already configured as 172.20.11.50, I just changed the Meadia termination point from ios based MTP to CUCM default MTP and the call connects

Thanks a lot it works, now I will go for the changes for whole organization. i have tested the pots call also,

[Ayodeji Okanlawon]

Glad to help. Don't forget to rate the posts!