similar to: new cisco VoIP phones

It would be nice if Digium updated the hardware compatibility list on asteriskpbx.org to indicate that the LineJACK can't be used for dialing out. I've seen several people on IRC be burned by not knowing this. --Eric -- BTEL Consulting 850-484-4535 x2111 (Office) 504-595-3916 x2111 (Experimental) 877-552-0838 (Backup Phone)

And it runs linux. http://www.zip4x4.com/ZIP4x4.htm Anyone seen one? bkw

Anyone having trouble parking calls? I haven't tried it in a while, but it seems to have stopped working. If I dial 700, I get a invalid extension. I have "include => parkedcalls" in the correct context, and I can dial 701, which tells me no call is parked there. Any ideas? Parking.conf is stock.

As I understand it (and my understanding is obviously incorrect) the switch => statement sells the Asterisk box to resolve (aka lookup) extensions by querying the remote Asterisk server defined in the switch => statement. The switch => statement is used to centralize dialplans. I've not used the switch => statement yet, I'm just trying to understand the ramifications of using

I've made some major changes to the way Asterisk handles file formats. I'd like feedback from people about any experience they have with these changes. They *may* improve playback performance for people who have had trouble with playback performance in the past. Mark

Maybe its just me but I find this question a little confusing, the tone duration should have no impact on tone recognition and typically in my experience the duration of the tone is defined by how long the user holds down the button !? > -----Original Message----- > From: James Sizemore [mailto:james@deny.org] > Sent: 22 August 2003 17:33 > To: asterisk-users@lists.digium.com >

I get the following message when I dial 74#. Does anyone have any ideas on what might be going on? If I don't require numbers to be terminated with # everything works as expected, but you have to wait for the digit timeout, of course. MESSEGE: DEBUG[1150520624]: File pbx.c, Line 1683 (ast_pbx_run): Oooh, got something to jump out with ('#')! -- Invalid extension '#' in

I'm getting the following message when I start Asterisk: WARNING[1024]: File codec_g729b.c, Line 413 (load_module): Unable to initialize va stuff: -1 Did I mess up the registration key or is something else wrong? --Eric -- BTEL Consulting 850-484-4535 x2111 (Office) 504-595-3916 x2111 (Experimental) 877-552-0838 (Backup Phone)

Licensing info for the G723.1 codec, direct from the holding company that licenses the codec. http://www.dspg.com/technology/LicensePricing.html As you can see they want a LOT of money. This is why I doubt there will ever be G.723.1 codec available fro Asterisk. -- BTEL Consulting 850-484-4535 x2111 (Office) 504-595-3916 x2111 (Experimental) 877-552-0838 (Backup Phone)

Is there any way on the iaxtel.com web site to see if my asterisk is registering and what 700 number is associated with my iaxtel account? I registered many months ago but never used it. My asterisk shows registered, but I can't seem to receive any calls (callers get a the user is not registered message) -- BTEL Consulting 850-484-4535 x2111 (Office) 504-595-3916 x2111 (Experimental)

I received a message from Packet8 last night telling me they added some new area codes. I took a look at their new area code/rate center finder page and it looks like the added a LOT of new area codes. They now have phone numbers available in almost every state. The URL for finding out which area codes and rate centers are available with Packet8 is at http://www.packet8.net/about/areacodes.asp

I use the following macro for my extensions. It only works with Zap channels and assumes that any Caller*ID number that is 4 digits is an internal call and all other calls are external calls. Use like this: exten => 1234,1,Macro(std-exten,Zap/4,20) [macro-std-exten] ; ; Caller*ID is 4 digits (internal call) ; exten => s/_XXXX,1,Dial(${ARG1}r2,${ARG2}) exten =>

Hi list, I would like to know of a possible way to dial a pstn number with an extension . Let the number is 56626965-234 so now i wanna dial 56636965 then wait for some time and dial the extension 234 to reach a particular person.I am afraid that i could not figure it out. I am trying in this way.. [outgoing] exten=>_566X.,1,wait,2 exten=>_566X.,2,Dial(${EXTEN})

Hi, I?m trying to get app_festival to work. I got the source from the Debian woody package of festival-1.4.2 and applied the patch that came with * sources it applied fine; then I made the debian package and installed it. I have this on extensions.conf: exten => 6700,1,Festival(Hi there how are you doing ?) When I dial 6700 I hear nothing and then * hangups: -- Executing

This is with an x100p (the motorola chipset) Two problems. Looks like CALLERIDNAME is being used uninitialized. On my other phones the callerid is fine and my buttset shows that the callerid passes the checksum. This is the relevant portion of extensions.conf exten => s,1,Answer exten => s,2,SetCallerID(H ${CALLERIDNAME} <${CALLERIDNUM}>) exten => s,2,Dial(${MGCP_ALL}) Here is

Hi there! Has anyone succesfully used a Cisco 12 SP+ with *? If so, how did you do? I'v not even tried, but before trying I thought I could bug you somewhat. =) //Filip

Am I the only person in the * world who can't get a sip connection through an iptables firewall? I've got everything else working fine. Xten <-> PSTN, Xten <-> Analog, IAX <-> IAX, but exten => 3733,1,Dial(SIP/fred@somewhere.com) ; evades me, ngrep @ port 5060 says the INVITES go out but how do I get something back? -- Dave Cotton <dcotton@linuxautrement.com>

Here's a patch that changes the behaviour of # transfers in asterisk. A single # is transferred to the remote phone/system. Two # in quick succession will trigger a transfer. This is very useful for users who have basic analogue phones connected to an ATA 186. For example, when calling a remote conference or IVR system you often want a single # to be sent to the remote system - not to

I'm not quite sure where to hunt for the answer to this. My apologies in advance if this isn't the appropriate forum. I have generally speaking used Kewlstart signaling on the POTS lines that I have used with asterisk, with great success. I have an installation where the POTS is coming from an in-house PBX (Panasonic TVS100) instead of the LEC, and on this one Kewlstart doesn't

Hi All, Is there any way to connect (register, initiate and receive calls) with Asterisk to FWD through NAT? Since I own my router port forwarding is not a problem. I tried with Register => <FWDnum>@fwd.pulver.com:<FWDpass>@fwd.pulver.com but since Asterisk still use internal IP in some SIP fields I got "479 We don't accept private IP contacts. Please set your external