07-01-2008 07:35 AM - edited 03-15-2019 11:39 AM
Hi all,
We have a setup where there are two CUCM6 publishers with a couple of phones registered to each cluster. Each CUCM has a SIP trunk configured to the same 2821 running a CUBE IOS. The configuration on the CUBE is:-
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
!
dial-peer voice 100 voip
description ** Outbound DP **
destination-pattern 9.T
session protocol sipv2
session target ipv4:10.212.1.101
dtmf-relay sip-notify rtp-nte
codec g711ulaw
no vad
!
dial-peer voice 200 voip
description ** Inbound DP **
session protocol sipv2
session target ipv4:10.212.1.103
incoming called-number 92051
dtmf-relay sip-notify rtp-nte
codec g711ulaw
no vad
All the phones on each cluster are G711. The problem is when a call is made from phone 123456 to 92051 the call rings through the SIP trunk across the dial peers, out through the other SIP and phone 2051 rings. However, there is only one way audio and when you examine each phone only 2051 has the G711 codec set as the receiver codec. On the other phone 123456 nothing is set for the sender or receiver codecs.
I cannot find out where the problem is. The device pools on the two CUCMs are identical so each phone has identical DPs, locations and regions, yet there is a mismatch somewhere.
Any ideas?
07-01-2008 12:43 PM
Try adding in voice service voip
SIP
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
sip
07-01-2008 12:45 PM
One other question, why would not just send the SIP request right to the CUCM servers? CUCM and CUCM BE will listen to SIP 5060 and 5061 by default. Send the SIP call directly to the server. I would think the CUCM would be able to pick this up and route it.
07-02-2008 12:20 AM
Thanks for the reply.
I'm trying to emulate a setup that we will be deploying elsewhere and wanted to get it working in the lab first.
I've added the SIP command to the config but it hasn't made a difference. The phones can ring so signalling is working. Then one phone will show the receiver codec as G711u and nothing else. The other phone does not show any codecs at all. So we get one way audio and then after about 20 seconds the call is disconnected.
07-02-2008 07:05 AM
OK have made some progress. Upgraded the cube to version c2800nm-adventerprisek9_ivs-mz.124-19b.bin.
Now I can successfully make voice calls across the CUBE with 2 way audio. However they only last for 18 seconds and then the call drops.
Any ideas?
Thanks,
07-04-2008 01:00 AM
07-04-2008 05:30 AM
Hi,
I have just simulated your scenarion in my lab. I got it work ok. However here is my suggestion as this is what I used in my lab...
Try this scenario..
CUCM6A --->h323 trunk --->CUBE ---> sip Dial Peer --->SIP TRUNK --->CUCM6B
CUCM6B --->h323 trunk --->CUBE ---> sip Dial Peer --->SIP TRUNK --->CUCM6B.
Summary:
CCM6A uses an h.323 trunk (ICT non-gatekeepeer trunk) to CUBE. From CUBE, sip dial-peers are configured to route the call to CCM6B. CCM6B is configured to accep the calls over a sip trunk.
CUBE config is as shown below:
Router#sh run
Building configuration...
Current configuration : 1473 bytes
!
version 12.4
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
!
hostname Router
!
boot-start-marker
boot-end-marker
!
!
no aaa new-model
!
resource policy
!
memory-size iomem 5
ip cef
!
!
!
!
!
!
!
!
!
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
h323
sip
!
!
!
!
!
!
!
!
!
!
!
!
!
!
!
!
!
!
!
!
!
!
!
!
!
interface FastEthernet0/0
ip address 172.16.5.44 255.255.255.0
duplex auto
speed auto
h323-gateway voip interface
!
interface FastEthernet0/1
no ip address
shutdown
duplex auto
speed auto
!
!
!
ip http server
no ip http secure-server
!
!
!
!
!
!
!
control-plane
!
!
!
!
!
!
!
dial-peer voice 102 voip
destination-pattern 30..
session protocol sipv2
session target ipv4:172.16.5.5
dtmf-relay rtp-nte digit-drop h245-alphanumeric
codec g711ulaw
!
dial-peer voice 108 voip
session protocol sipv2
incoming called-number 40..
dtmf-relay rtp-nte
codec g711ulaw
!
dial-peer voice 109 voip
session protocol sipv2
incoming called-number 30..
dtmf-relay rtp-nte
codec g711ulaw
!
dial-peer voice 103 voip
destination-pattern 40..
session protocol sipv2
session target ipv4:172.16.5.6
dtmf-relay rtp-nte digit-drop h245-alphanumeric
codec g711ulaw
!
!
!
!
!
gatekeeper
shutdown
!
!
line con 0
line aux 0
line vty 0 4
login
!
!
end
Router#
NB: DN 40.. is my CCM6B
DN 30.. is my CCM6A.
I configured two trunks each on both ccm..one is an h.323 trunk for outgoing calls...the other is a Sip trunk for incoming calls...
Make sure that both trunks point to your CUBE gateway
NB: you have to use software MTP on both trunks or on atleast one of the trunks..
So ensure the trunk has a MRG assigned to it and check the MTP required box..
Hope this helps.
For troubleshooting Incoming SIP calls..
enable debug ccsip messages
debug ccsip events
debug ccsip calls
07-04-2008 06:54 AM
Thank you very much for taking the time to help.
I have tried your configuration and it works perfectly!
I guess that what I was trying to achieve CUCM6A --> SIP --> CUBE --> VOIP DP --> SIP --> CUCM6B
is just not possible.
Many thanks.
07-08-2008 08:24 AM
Hi, looking into aokanlawon's configuration, I have a question.
For the scenario is
CUCM6A --->h323 trunk --->CUBE ---> sip Dial Peer --->SIP TRUNK --->CUCM6B
You are using 108 as incoming and 103 as outgoing dialpeer, right?
I don't know why you are using an ICT but the 108 dialpeer still have "session protocol sipv2" in it.
Could you please explain? Thanks.
07-08-2008 08:30 AM
yes, 108 is incoming and 103 is outgoing..
without the session protocol sipv2 in the incoming, the h.323 trunk coming from callmanager does not get "converted to sip" which is what we use for outgoing...hence if that is not there, the call will proceed with h.323 all the way and the incoming trunk on the CUCM6B is a SIP trunk..
NB: You can also use a sip to sip trunk all the way or use an h.323 to h.323 trunk all the way.
07-08-2008 09:38 AM
not necessarily. I have used H323 from CUCM to the router, then SIP to another end point and it was fine. I did have a problem with the router IOS version though. I had to upgrade to latest and greatest to get SIP to work correctly on the router. The router can handle the H323 to SIP fine, it's the IOS version that can not handle it.
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