search for: analogphone

On one installation of asterisk, I have a display on the console when I have a incoming call on my zaptel card. every digit was displayed, this was great. Does anyone know how I can get this back? Thanks

...told "701" The extension I'm dialing goes to the on hold music. I'm disconnected, I hang up, dial "701" and I see this message on the console "Everyone is busy/congested at this time" I just have the default parkedcalls file, and have this in the extensions. [AnalogPhone] exten => _70X,1,Dial(Zap/1/${EXTEN},20,Ttr) include => parkedcalls [SipPhone] exten => _70X,1,Dial(SIP/1/${EXTEN},20,Ttr) include => parkedcalls

Hi, I have 2 asterisk boxes as Gateway, in this arrangement. (PANASONIC PBX) - [ASTERISK1] - network - [ASTERISK2] - (ANALOG PHONE) everything works great, in both directions (receiving and making calls), but when i get a call on the (ANALOGPHONE), I haven't been able to transfer it to another extension of the PANASONIC PBX using the flash key. I've tried the using the t T options on the 'Dial(' command of the extensions, but no luck. I assume that's more for transfering on the asterisk domain. I just want to pass th...

...stDB. Advice is apreciated, can't seem to find an answer. ; define channels group=1 context=longdistance_users signalling=fxo_ks ;FXO Sig for Phone callerid="John French" <103> mailbox="101" callwaiting=yes threewaycalling=yes transfer=yes channel => 1 setvar=USER=analogPhone

...sk-users@lists.digium.com > Subject: [Asterisk-Users] DTMF Detection Problem > Reply-To: asterisk-users@lists.digium.com > > --0-1376241818-1080711919=:58147 > Content-Type: text/plain; charset=us-ascii > > Hi, > My set up is like this > Asterisk--->SipuraATA----->AnalogPhone > When I'm calling into asterisk from a cell phone, there's no dtmf detection problem as asterisk can detect correct extensions that I press. But when the phone is further connected to the AnalogPhone thru the ATA, the dtmf signal is really short/weak. I've tried to adjust dtmf lengt...

...-- Hungup 'Zap/1-1' Here is my incoming extensions.conf dialplan: [globals] FWDNUMBER=223611 ; your calling number FWDCIDNAME="Robert Webb"; your caller id FWDPASSWORD=password ; your password FWDRINGS=IAX2/rwebb ; the phone to ring FWDVMBOX=2002 ; the VM box for this user ANALOGPHONE=zap/2 OFFICEPHONE=SIP/2002 LAPTOPPHONE=IAX2/rwebb VMBOX=2000 PSTNOUT=zap/5 [fromPSTN] exten => s,1,LookupBlacklist exten => s,2,DigitTimeout(3) exten => s,3,ResponseTimeout(5) exten => s,4,Wait(1) exten => s,5,Background(custom/pls-wait) exten => s,102,Goto(blacklisted,s,1) e...

...AX2/xxxxxx@NuFone/${EXTEN} I have a couple of Xlite softphones and 2 analogue phones connected to a mediatrix 1102 connected to our lan. The mediatrix talks sip to the asterisk box on the lan. We are running asterisk on FC3 . SOFTPHONES[XLITE] ---SIP--> ASTERISK----IAX--->NUFONE[ASTERISK] ANALOGPHONES---MEDIATRIX_1102---SIP--->ASTERISK---IAX--->NUFONE[ASTERISK] Well the problem goes something like this. 1) I can dial a number form the softphones and when the call is answered I can hear the user on the other end but the user can't hear me 2) I can dial a number from the analog phones...

...sk-users@lists.digium.com > Subject: [Asterisk-Users] DTMF Detection Problem > Reply-To: asterisk-users@lists.digium.com > > --0-1376241818-1080711919=:58147 > Content-Type: text/plain; charset=us-ascii > > Hi, > My set up is like this > Asterisk--->SipuraATA----->AnalogPhone > When I'm calling into asterisk from a cell phone, there's no dtmf detection problem as asterisk can detect correct extensions that I press. But when the phone is further connected to the AnalogPhone thru the ATA, the dtmf signal is really short/weak. I've tried to adjust dtmf lengt...

...versation stop hearing each other. I can hear that they cant hear each other. We are using chanspy and passing the exact session-name to it as an argument. Asterisk 1.6.0.6 is being used a B2BUA. It receives calls from users connected via analog phones to a Quintum and forwards them to a PSTNGW. Analogphone--Quintim<------ SIP------->Asterisk<-----SIP------>PSTNGW(T1s) All media is passing thru asterisk (canreinvite = no). Thanks in advance for any help -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attach...

...ECT working? I can't get it to work - please see my logfiles below. While using an * CVS version of late September and chan_capi-0.2.5 (I guess), it worked!!! (I know, never change a running system ... should've backuped ... etc.) Here's the setup: NT----AgfeoISDNPBX----AgfeoISDNPBX----AnalogPhones in between the two PBX I attached my * server with a FritzPCI card to the S0-bus. I have some Homeautomation on the server and * is supposed to route the calls depending on certain conditions. Switching an incoming call using * "Dial"-Feature would block the two internal B-channels, that...

...AX2/xxxxxx@NuFone/${EXTEN} I have a couple of Xlite softphones and 2 analogue phones connected to a mediatrix 1102 connected to our lan. The mediatrix talks sip to the asterisk box on the lan. We are running asterisk on FC3 . SOFTPHONES[XLITE] ---SIP--> ASTERISK----IAX--->NUFONE[ASTERISK] ANALOGPHONES---MEDIATRIX_1102---SIP--->ASTERISK---IAX--->NUFONE[ASTERISK ] Well the problem goes something like this. 1) I can dial a number form the softphones and when the call is answered I can hear the user on the other end but the user can't hear me 2) I can dial a number from the analog phones...

Hi, I am sorry to be asking this but the wiki is down and has been for a couple of days and I need to get this working before Monday to get my live system setup. Trying to get the Snom 190's and soon to arrive 3com 3102's to use the function keys and for the life of me I can't work it out from the conversations on the archive what I am going exactly wrong here? The snom 190 with