search for: call_wait

...ond call to be able to come in, otherwise the phone returns a Busy here, while with the older versions it could have been disabled and it worked fine, the phone only returned busy here on the 7th call. So I had to enable call waiting, the way I did it was that in the SIP<mac>.cfg file I added call_waiting: 3 I'm not sure if this is related or not, but that was the only change I had to do to the config files.

...nymous Call Blocking (0-disbaled, 1-enabled, 2-disabled no user control, 3-enabled no user control) anonymous_call_block: "0" ; Default 0 (Disable blocking of anonymous calls) # Call Waiting (0-disabled, 1-enabled, 2-disabled with no user control, 3-enabled with no user control) call_waiting: "1" ; Default 1 (Call Waiting enabled) # DTMF AVT Payload (Dynamic payload range for AVT tones - 96-127) dtmf_avt_payload: "101" ; Default 100 # XML file that specifies the dialplan desired dial_template: "dialplan" # Network Media T...

I have a 7960 with P0S3-08-2-00 firmware. Call waiting is enabled on the phone: ... call_waiting : 1 ... When I call from the 7960 to the 7960, I get a 486 busy here on the asterisk console. I can call from the 7960 to a soft phone no problem and vice-verse. I can also call from soft phone to soft phone with no issues either. Anyone know offhand what's going on? bug in the firm...

...on the phone when another call comes in, there is about a 10 second time during which the user can't hear the call they're on. Then it returns to normal. My research indicated that I needed to add these setting to the phone: ======================================================== <CALL_WAITING tone.chord.callProg.6.freq.1="440" tone.chord.callProg.6.level.1="-19" tone.chord.callProg.6.onDur="300" tone.chord.callProg.6.offDur="100" tone.chord.callProg.6.repeat="2"/> <callWaiting call.callWaiting.ring="ring"/> reg.1...

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

I'm not sure if this is an Asterisk thing, a handset thing or a telco thing, so please be gentle with me if this is not the right place to ask ..... When placing a call over a SIP channel to a mobile phone, if the phone is engaged, it does not return a BUSY status straightaway. Rather, I get a ringing-out tone for the timeout duration specified in the Dial() statement; *then* I get

...nymous Call Blocking (0-disbaled, 1-enabled, 2-disabled no user control, 3-enabled no user control) anonymous_call_block: "0" ; Default 0 (Disable blocking of anonymous calls) # Call Waiting (0-disabled, 1-enabled, 2-disabled with no user control, 3-enabled with no user control) call_waiting: "1" ; Default 1 (Call Waiting enabled) # DTMF AVT Payload (Dynamic payload range for AVT tones - 96-127) dtmf_avt_payload: "101" ; Default 100 # XML file that specifies the dialplan desired dial_template: "dialplan" # Network Media Typ...

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

Anyone have a Snom they're happy with? How did you manage that? :) I have a system of: Asterisk 1.2.3 2 Wildcard TDM400P Rev I and E/F 1 Snom 360 + sidecar ~15 Sipura/Linsys SPA-841 ~15 Grandstream 101 Everything (currently) is on the same network, not a router to be seen between any two. Also everything, except the snom, is working sweetly. The main problem is ECHO.. awful echo and