search for: dtmf_outofband

...ll allow=alaw allow=ulaw callgroup=2 pickupgroup=2 and in the SIPDefault.cnf for the phones I have: # Inband DTMF Settings (0-disable, 1-enable (default)) dtmf_inband: 1 # Out of band DTMF Settings (none-disable, avt-avt enable (default), avt_always - always avt ) dtmf_outofband: avt # DTMF dB Level Settings (1-6dB down, 2-3db down, 3-nominal (default), 4-3db up, 5-6dB up) dtmf_db_level: 3 DTMF works for voicemail and for remote services over both analogue Zap channels and digital (ISDN) channels. Asterisk doesn't appear to be 'monitoring' the audio...

...he ability to telnet into this phone telnet_level: "2" ; 0-Disabled (default), 1-Enabled, 2-Privileged # Inband DTMF Settings (0-disable, 1-enable (default)) dtmf_inband: "1" # Out of band DTMF Settings (none-disable, avt-avt enable (default), avt_always - always avt ) dtmf_outofband: "avt" # DTMF dB Level Settings (1-6dB down, 2-3db down, 3-nominal (default), 4-3db up, 5-6dB up) dtmf_db_level: "3" # SIP Timers timer_t1: "500" ; Default 500 msec timer_t2: "4000" ; Default 4 sec sip_retx: "10&quo...

I have noticed a problem with the Cisco 7940/7960 phones where if you put your voice-mail box on hold using soft keys and come back you can no longer navigate. I am curious if anyone else can duplicate this problem. Happens reliably for me with the 7940 phones. I use rfc2833 for DTMF. I would think it was a Cisco bug, but for the fact that this did not happen with older version of

...RTP playload 101 has been sent to the cisco phone, but no audio. in the dialplan exten => 8603,1,Answer(1) exten => 8603,n,sipdtmfmode(rfc2833) exten => 8603,n,SendDTMF(1|100) exten => 8603,n,hangup() sip.conf dtmfmode=rfc2833 SIPDefault.conf I did play with all possible settings for dtmf_outofband: avt, avt_always, none and 0,1 for dtmf_inband nothing happens cisco 7905g is working OK with this example cisco 7960 firmware issue? Any help?

Hello all. We have an asterisk system set up, and we are seeing a lot of multiple DTMF digits being read by asterisk. In digging through the archives the only answer I have seen is to put in the statement relaxdtmf=yes in the zapata.conf file. Since we are not using any zapata devices, I have tried to put that statement in my sip.conf file to no avail. Any help would be appreciated as my end

Anyone have experience with the 3-08-2 release of Cisco's SIP firmware? Are there any new features in the SIPDefault.cnf? Thanks, Ron

...the ability to telnet into this phone telnet_level: "1" ; 0-Disabled (default), 1-Enabled, 2-Privileged # Inband DTMF Settings (0-disable, 1-enable (default)) dtmf_inband: "1" # Out of band DTMF Settings (none-disable, avt-avt enable (default), avt_always - always avt ) dtmf_outofband: "avt" # DTMF dB Level Settings (1-6dB down, 2-3db down, 3-nominal (default), 4-3db up, 5-6dB up) dtmf_db_level: "3" # SIP Timers timer_t1: "500" ; Default 500 msec timer_t2: "4000" ; Default 4 sec sip_retx: "10&quo...

...eam (g711ulaw (default), g711alaw, g729a) preferred_codec: g711ulaw # TOS bits in media stream [0-5] (Default - 5) tos_media: 5 # Inband DTMF Settings (0-disable, 1-enable (default)) dtmf_inband: 1 # Out of band DTMF Settings (none-disable, avt-avt enable (default), avt_always - always avt ) dtmf_outofband: avt # DTMF dB Level Settings (1-6dB down, 2-3db down, 3-nominal (default), 4-3db up, 5-6dB up) dtmf_db_level: 3 # SIP Timers timer_t1: 500 ; Default 500 msec timer_t2: 4000 ; Default 4 sec sip_retx: 10 ; Default 10 sip_invite_retx: 6 ; Default 6 timer_invite_expires: 180 ; Default 180 sec...

...dia stream (g711ulaw (default), g711alaw, g729a) preferred_codec: none # TOS bits in media stream [0-5] (Default - 5) tos_media: 5 # Inband DTMF Settings (0-disable, 1-enable (default)) dtmf_inband: 1 # Out of band DTMF Settings (none-disable, avt-avt enable (default), avt_always - always avt ) dtmf_outofband: avt # DTMF dB Level Settings (1-6dB down, 2-3db down, 3-nominal (default), 4-3db up, 5-6dB up) dtmf_db_level: 3 # SIP Timers timer_t1: 500 ; Default 500 msec timer_t2: 4000 ; Default 4 sec sip_retx: 10 ; Default 10 sip_invite_retx: 6 ; Default 6 timer_...

Hi everyone, I've been toying with * for quite some time now. I've got two Cisco 7940's with the SIP firmware playing nice with *. I can also make outbound calls via IAXTel (toll-free calls only) and all other calls I have routed out my X100P-clone adapter. Here's my question... Is there a way to capture the inbound callerid from my phone line (coming in on the X100P) and have

Thanks But how about the config files (SIP...) that needs to be inside the tftp server, where can I get a sample of that? That's where the images for the firmwares of the ip phones come from, on boot right? -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Alen Salamun Sent: Wednesday, January 26, 2005 5:47