search for: dmtfmode

I'm dialing DTMF's in a SIP channel using the options: [sip.conf] dmtfmode=info [extensions.conf] exten => _XXXXXXX,1,Dial(SIP/gateway,,D(${EXTEN})) (this is a custom SIP gateway, which receives the DTMF's sent from softphones through Asterisk, and based on them, build the destination PSTN number). My problem is that Dial send the DTMF's to the SIP/gateway...

...age=fr useragent=Asterisk PBX dtmfmode = auto register => 09XXXXXXXX:SECRET@freephonie.net registertimeout=40 externip = 82.XXX.XXX.XXX localnet=10.XXX.XXX.XXX/255.255.255.0 qualify=60000 nat = yes [test] type=friend username=test secret=test host=dynamic context=home callerid =test <2222> dmtfmode=rfc2833 authuser=test fromuser=test allow=all [freephonie_outbound] type=peer allow=all host=freephonie.net secret=SECRET fromuser=09XXXXXXX username=09XXXXXXX dtmfmode=inband qualify=60000 fromdomain=freephonie.net [freephonie_inbound] type=peer context=incoming host=freephonie.net qualify=60000 a...

Hi, I searched the archives about this, but didn't find any references. When I make an outbound SIP call, the call completes and everything is fine, but in the Asterisk console, I keep getting a huge stream of warning messages: "WARNING[1200876848]: File dsp.c, Line 1107 (ast_dsp_process): Unable to detect process 2 frames" I thought I saw this in a post earlier, but I

Hello, I have one problem and I need to disable dtmf (disable rfc2833, info and inband) on one (other peers must support dtmf) SIP peer . Is it possible? Workaround would be use g729 codec with dtmfmode=inband. Maybe there is better solution? Thanks for help. -- Pagarbiai / Best Regards, Giedrius Augys -------------- next part -------------- An HTML attachment was scrubbed... URL: