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New Member

Reg. H.225.0 CS:Release Complete


I have the following network arrangement

Cisco 7940 Skinny Phones (no: 1133) <-> Cisco Call Manager <-> Cisco 2611XM router (IP-IP gateway) <-> SIPH323 converter <-> SIP Proxy <-> Cisco 7940 SIP Phone (3322).

When I call from SIP to Skinny phone, the skinny phone rings. When I take the hook of Skinny phone, it gives me busy tone and so also in SIP phone. I traced the message using ethereal and I could see that Cisco 2611XM router is sending a H.225.0 CS: Release Complete message at some point. The real trace communication goes like this

router CM

|<-----------| H.225.0 cs: Connect

|----------->| H.245 TerminalCapabiltySet

|----------->| H.245 MasterSlave Determination

|<-----------| H.245 TerminalCpabilitySet

|<-----------| H.245 TerminalCapability SetAct

|----------->| H.225.0 CS: ReleaseComplete

|----------->| H.225.0 CS: ReleaseComplete

It would be great if anybody would tell me what I have to configure in my router to avoid this problem. I have the following

configuration in my router.

......default conf


voice service voip

no allow-connections any to pots

no allow-connections pots to any

allow-connections h323 to h323

dial-peer voice 300 voip

dest. patt. 1133

ses. tar.. CM IP address

dial-peer voice 301 voip

dest. patt. 3322

ses. tar.. SIPH323IP address

I would appreciate if anybody could suggest me a solution for this.

Thanks in advance,

Balaji Thoguluva


Re: Reg. H.225.0 CS:Release Complete

Chances are the IP phones are tring to use a G711 codec and the IP to IP gateway is attempting to use the default G729 codec, so the call fails on the h245 caps exchange. Try adding codec g711ulaw commands on the VOIP dial peers of the IP to IP gateway.

New Member

Re: Reg. H.225.0 CS:Release Complete

I added codec command in IP to IP gateway dial-peer configuartion and I made sure the Callmanager and cisco router use the same coding scheme. Even then the problem exists. I would appreciate if you would tell me what might be the reason and how to solve it.



New Member

Re: Reg. H.225.0 CS:Release Complete

Try this:

dial-peer voice 3 voip

codec transparent


Service Parameters -> CCM ->

-> SendH225UserInfoMessage -> H225InfoforRingBack

But then no ring back if call transfer to local phone???