similar to: Delayed ringing on some SIP phones

Hello In my extensions.conf file: [frompstnisdn] exten => s,1,Dial(SIP/200&SIP/202,20) exten => s,2,Voicemail(su200) exten => s,3,Hangup I use the s, start, extension to handle incoming calls. In my zapata.conf: context=frompstnisdn This works ok on another asterisk box I setup. But on incoming calls I get: -- Extension '787367' in context 'frompstnisdn'

I would like to setup a system with two tennants with two seperate DIDs through one IAX provider account. Is it possible to route the calls into different contexts based on the DID dialed? I have searched and found nothing. I do not see anywhere in the console that says what DID was dialed so I am thinking two seperate accounts are needed to make this work. Can anyone confirm? Thanks

Hi All, I Have a Digium Card TDM40P, the specification say: OEM TDM40B: TDM400P + 4 PORT FXS Bundle, my question is: Can I to install analog lines of PTSN?, other detail is: this card have 4 card green. I need to know what is the best card for the following scenario: I need a IVR for my comapny and a PBX, but i want that my extension not use FXS I want IP phone . Thanks ins advanced,

So, I got my wife a Nook for her b'day. I just plugged it into my system, CentOS 6.5, and what I see is /media/NOOK, and it shows 257k or so - yes, k, not m or g. It's *not* seeing any directories, etc, and the small thing I'm guessing is firmware, since even when I try mount -o remount -rw, it is still r/o. Googling, I see mentions of fsmtp, I think it was, and yum shows some mtp

On Wed, Jan 27, 2016 at 02:30:15PM +0000, Timothy Murphy wrote: > Jonathan Billings wrote: > > >> > Maybe you're not > >> > aware of it, but there are a LOT of things that systemd fixes that > >> > people are happy about. > > >> Like what ? I don't remember there were as many errors to fix before > >> systemd appeared. >

Is there any way to take an incoming callerid string and remove the given "name" part of it and replace it w/ something arbitrary, or add to a blank name string (possibly by looking up the number in a database)? Thanks, John Lawler

Hi, Does anybody know of a VoIP compatible doorbell or door intercom unit? I've contemplated buying a cheap SIP phone, ripping it apart, and putting it inside an IP66 sealed unit... It would need: - At least one speed-dial key, or some way to make every button dial the same extension number - PoE (power over ethernet), so I can power it off the central switch - cheap enough to rip apart

When somebody call me on my pstn # cable connected to my fxo card it does not work when I check my computer the following error shows Connected to Asterisk CVS-v1-0-12/05/04-19:46:25 currently running on asterisk1 (pid = 2160) Verbosity is atleast 3 -- Remote UNIX connection -- Starting simple switch on 'Zap/1-1' == Starting Zap/1-1 at incoming,s,1 failed so falling

I've got a need to do something like the following: [foo-context] exten => _.,1,SetCIDNum(123) exten => _.,2,SetCIDName(XYZ) include => local include => tollfree But of course, this example won't work. The goal here is this: if a call ends up being handled by the "local" or "tollfree" contexts, I want those SetCID*** commands executed. Otherwise, I

Hi to all, to manage properly a call center for multiple companies is possible to let the X-lite/X-Pro softphone to display the number or context called from PSTN to let operator answer with the correct name of the company?? I explain better. If a call come from PSTN to Number A for company A i want the operator recognize it and answer "Good Morning, I'm Operator of company A"

Chaps, I recently added an incoming VOIP account to my asterisk box. When the PSTN number associated with this account is dialled, the call rings once and then asterisk starts playing music on hold, even though all the extensions continue to ring. Variations of answer() and ringing() don't seem to help. I'm sure I'm missing something spectacularly obvious, but the wiki and googling the

Hello: I have read rsync manual and several howtos on how to use rsync, still I don't know if it's doable what I want to do, and if yes, how. The scenario: I would like to make a local copy of openSUSE 13.2 repositories. I use openSUSE linux. The repos are located in a multi-level directory structure, eg: ftp://ftp.halifax.rwth-aachen.de/opensuse/repositories/ Let's call this

I am experiencing a problem booting a machine with ISOLINUX on a machine that has no hard drive (CD-ROM only). I've tried two different machines, a Dell Poweredge 2600 and a Dell Poweredge 1300, both exhibit identical symptoms. When I boot the exact same cd on a system with a hard drive, the boot is just fine. I had been using ISOLINUX 1.66, but I just upgraded to 2.02 and I am having the

Hi I would like to send a text to the called person when he picks up the phone before the call gets connected through. Is there a way to do this? Example: I'm registered to multiple SIP providers. They come in to a context each and then get through to my phone. Now I would like to send myself an announcement about from which SIP provider this call came from. -- Beno?t Panizzon,

Folks, I'd like to change the value of ${CALLERIDNAME} for incoming PSTN calls from certain numbers, but haven't found a way that works. The goal is to provide more informative names on my phones' caller ID displays--e.g., I would prefer to display "ROB CELL" instead of "CELLULAR CALL" when I call home from my cell phone. This is what I tried in the context

Does VoicemailMan have to be installed ? Why not available. I have setup a mailbox in voicemail.conf and I can leave a voicemail - just cannot pickup up using *97. My *97 code in extensions.conf: exten => *97,1,Answer exten => *97,2,VoicemailMain(${CALLERIDNUM}@default) exten => *97,3,Hangup asterisk console: Verbosity was 8 and is now 12 -- Executing

I am trying to get a ROR app running on a linux box running Apache 1.3 at my hosting comapny, but cannot get it running. There is a fresh/clean ROR app in the base directory / .htaccess, dispatch.cgi, dispatch.fcgi and dispatch.rb are all in / All dispatch files have been set to 755. ROR docs say to point doc root at /public/ and to put .htaccess, dispatch.cgi, dispatch.fcgi and dispatch.rb

Hi All, * is working very well for us now. But I have an issue that I cannot find the answer to - enter guru's!! When our receptionist does a blind call forward I receive the Caller ID, however I do not know if the call is fresh (i.e. ringing in) or forwarded. What I would like to do is to have * prefix the CID External (so that I can tell that it is a fresh call) or Internal (to tell me

If anyone could help me with this, I'd appreciate it! I've got an Asterisk deployment where I'd like to use an existing external Octel voicemail system. I've been trying to define an extension that if the call isn't answered in a few rings, to dial our external voicemail number. That voicemail system works by seeing the CALLED number and routing the call to the

was wondering if anyone could give us a run through an explanation of the wiki and other examples of connecting to iptel's sip express router using asterisk pbx so i can understand better the call processing .. given the example i work from on john todd's www.loligo.com site ; exten => _3.,1,SetCallerID(${IPTELUSERID}) exten => _3.,2,SetCIDname(${IPTELUSERNAME}) exten =>