23 Jan
2004
23 Jan
'04
10:14 p.m.
On 23-01 14:27, Balaji Bapulal Thoguluva wrote:
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).
This is crazy, do you really need all the protocols and protocol
conversions ?
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.
No, you can not do this with ser.
Jan.