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 > [EMAIL PROTECTED] > 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 > > > > _______________________________________________ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users
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