Len
2008-Jan-07 11:57 UTC
[asterisk-users] Strange migration problems from asterisk 1.2.13 to 1.4.10, dtmf related?
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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080107/34961ab2/attachment.htm
Len
2008-Jan-08 11:15 UTC
[asterisk-users] Strange migration problems from asterisk 1.2.13 to 1.4.10, dtmf related?
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 > > > > _______________________________________________ > --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-------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080108/8a5ec3ce/attachment.htm