[Serusers] Reg. Codec Setting in SER for SIP Phones

[Balaji Bapulal Thoguluva]

Hi,

   I have explained my problem below and I have given my question related to SER at the end. Any suggestion for the question would be of great help to me. Also I would appreciate if there is any solution for the problem I have described.  

   I have the following network.

Cisco SIP Phones (3322) <-> SIP Express Router (SIP Proxy router) <-> SIPH323 Converter <-> Cisco router 2611XM (acting as IP to IP Gateway) <-> Cisco CallManager <-> Cisco Skinny Phones (1133).


First Problem (SIP->Skinny):
---------------------------
   When I call from SIP phone to skinny phone, the skinny phone rings. But when I take the hook, it gives me busy tone. I see from ethereal that the router is sending H.225.0 cs: Release Complete message to CallManager. My router dial-peer config is

dial-peer voice 300 voip
   dest pattern 1133
   session target CM's IP address
   codec g711alaw

and my CallManager configuartion is: I use my default device pool that has default region using codec g711. I could also see that cisco skinny IP phone config. in
CallManager uses default device pool defined above.

Second problem (Skinny->SIP):
-----------------------------
        The same network configuartion is assumed. When I call from Skinny to SIP phone, the SIP phone just rings once. The SIP phone doesn't seem to ring continuosly (could not hear the dail tone) but shows a missed call from skinny phone.

        When I traced the call flow using ethereal, I see again the following

       router       CallManager
         |----------->| H.225.0 cs:Alerting
         |<-----------| H.245 TerminalCapabiltySet
         |----------->| H.245 Terminal CapabiltySet
         |----------->| H.245 MasterSlaveDetermination
         |<-----------| H.245 TerminalCapabaility Ack
         |----------->| H.225.0 cs:Release Complete

My dial-peer config is

dial-peer voice 201 voip
    dest patt 3322
    sess tar. SIP Proxy's IP
    codec g711alaw

    So, the conclusion is in both cases the router is sending Release complete message to CallManager. So I guess there is some capability mismatch between router and CallManeger. I guess SIPH323 is flexible to use any codec.
 
Question: I have a slight doubt that is there anyway I can set the codec used by the sip phones in SER router. If there is any way, please throw some light on this issue. 
     
Thanks,
Balaji