Re: [asterisk-users] Strange migration problems from asterisk 1.2.13 to 1.4.10, dtmf related?

2008-01-08 Thread Len
Hello again, Just to close this I have found the problem to be related to 1.4.10. For some unknown reason the sip debug showed Found description format PCMU for ID 0 Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec

[asterisk-users] Strange migration problems from asterisk 1.2.13 to 1.4.10, dtmf related?

2008-01-07 Thread Len
Hello, I have the following problem. I am migrating my asterisk infrastructure to a new server and I encounter a strange problem. The configuration is as followin: IAX clients connect to asterisk which forward calls to a sip box connected to a phone line. On the old server everything works ok but