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

[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 capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing)


after upgrading to 1.4.17 everything worked ok again with the same
configuration files:

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 capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)


All here:
http://www.len.ro/work/tools/gutsy-on-a-ubuntu-server/asterisk/view

Best regards,
Len
http://www.len.ro




On Mon, 2008-01-07 at 13:57 +0200, Len wrote:

> 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 on the new server, even if the logs are
> identical it seems like the dtmf number does not get passed correctly
> to the sip box as the phone does not dial the proper number. The log
> shows something similar to:
> 
> [Jan  7 13:33:11] VERBOSE[7785] logger.c:     -- Called 1002
> [Jan  7 13:33:11] VERBOSE[7785] logger.c:     -- SIP/1002-081b4a80
> answered IAX2/ioper00-1
> [Jan  7 13:33:11] VERBOSE[7785] logger.c:     -- Sending DTMF
> 'w0214108658' to the called party.
> 
> where 1002 is the sip box
> 
> [1002]
> type=friend
> username=1002 at 10.0.0.1
> callerid="1002"
> secret=xxxxxxx
> host=dynamic
> dtmfmode=inband
> deny=0.0.0.0/0.0.0.0
> permit=10.0.0.121/255.255.255.255
> 
> The only problem I can think of is dtmf related. Did something change
> from asterisk 1.2.13 to 1.4.10 which could cause this problem? Can it
> be related to the computer speed (very unlikely in my mind).
> 
> Thank you very much for any ideeas as I am bumping my head for a hole
> day trying various combination.
> 
> Best regards,
> Len
> http://www.len.ro
> 
> 
> 
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