similar to: Caller ID forwarding

I've got the following set up: Someone calls into my PBX on a single number (via SIP trunk from my carrier), and the get a voice menu of extensions. On one of the extensions, it rings a bunch of internal SIP hardphones, plus ringing my cellphone via a hairpin through the cairrier's SIP/PSTN gateway. The issue is that my cellphone shows my PBX's number, not the original calling

Hello all, I want to setup Asterisk to forward a call if the dialed extension is busy. I do not want to wait on the line until the extension timeout expired. What I want is when I dial am extension currently Busy (Talking with someone), asterisk inmediately forwards my call to an extension I previosly defined. Someone could help me? Any clue will be appreciated. Regards from Spain. Ismael

Hi! Maybe I have just been looking on the wrong pages but there is a question that is very important for me. I already studied some Demo-Dialplans and made some basic experiences with Asterisk. But what I need to find out is how I can handle this. I am leaving my office and I want to tell asterisk to forward calls now to my mobile phone by just hitting a key (on my IP-Phone) or by using a

I'm working on building a pbx that will allow us to use our cellphones as extensions (to some extent) The dialout is working fine. What I would like to do is have an inbound cellphone call appear as if it were an extension. So right now if I call in from cell #9995551212 the caller id is 9995551212 but if I dial extension 30013 it will call cell #9995551212. I would like to change the caller

Hi; I've been using Asterisk for a few months now, and I have run into an interesting issue that I thought someone else in the community may have run into: I have an Asterisk install set up to receive helpdesk calls, route them to several IAX extensions and an extension which is simply a forwarded call over the POTS to a cellphone, so that if no one is logged into their IAX extensions for

I've tried to setup multiple extension dialing - ie dial 1 number and it rings at a number of sources. For the most part its worked.... Now if someone dials 107 it rings Sip phones at 102 and 107, then goes to voicemail after 40 seconds. exten => 107,1,Dial(SIP/102&SIP/107,40|r) exten => 107,2,Voicemail(u102@pstn) exten => 107,3,Hangup exten => 107,102,Voicemail(b102@pstn)

Is there any way we can make use of the call forwarding feature on our Telco phone line. I've seen this question asked on this list before but looking in the archive i don't see that it has been answered. If someone has this working or knows how please let me know. Thanks.

I never did see this get to the list. Tim Litwiller wrote: > For my church school we need a way to connect 3 room phones, 1 office > phone and 2 phone lines. > so I need a device or several that i can connect to 2 pots phone lines > and at least 3 plain old wall phones. I'll donate a sipura 941 for > the office. > > What would be the best product to get 2 fxo ports

Hi list, This is not necessarily an asterisk issue, but a lot of you guys know way more then me, so I have a question: someone at my company sets his phone to forward calls to his cellphone, so someone calls our office, call is forwarded to his cell, and the callerID that shows up on his cell is of course our office number, because asterisk originates a new call to his cell and then bridges

I haven't found this in any docs or faqs yet, so I'm wondering if I can achieve what I would like to do. On an Asterisk PBX with multiple PSTN lines, I'd like to call in from one PSTN line, probably via cellphone and access the PBX as if I were local to it. From here I'd like to get a dial tone and call back out. I know this isn't exactly call forwarding per se, but I'm

I have a customer looking for an automated way to provide his customers information. He found some windows software called Active Call Center - but I believe that he did the 40 day trial and it crashed his windows machine to much. So he wants something that can do a similar task running on linux. The Users expierience should go something like this One of his customers calls in the IVR asks

I've had this discussion in the office and with some vendors, but no one has a solid answer, hopefully someone here does. What is the proper way to format a caller-ID here in the U.S.? Is it: +15705551212 or is it +5705551212 I've always seen it +15705551212, but as I understand it the country code for the US is 011, which to me would indicate you put 011-570-555-1212 as the callback

Hi How do i set up so that everyone can dial, for example *21* to forward all calls to a cellphone or another extension and how do I enable so that cals can be transferd between extentions. I use asterisk 1.6 and have my phones in unistim.conf and my extensions in extensions.conf. Regards /ralf ________________________________________________ Ralf Tr?skman, IT AdLibris AB, Odengatan 106, 113

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I have an iaxy the is across a 802.11b link from my asterisk server. signal strength is good and it has been working fine there for about a year. Friday night lightning took out the x100p card in my asterisk server and I just got it all working again last night. Lucky I had a spare. Since I got every thing setup I have been having problems with the iaxy - I really doubt that it is lightning

*@home has this for it's incoming fax macro --- start snip --- [ext-fax] exten => in_fax,1,GotoIf($[${FAX_RX} = system]?2:analog_fax,1) exten => in_fax,2,Macro(faxreceive) exten => in_fax,3,system(tiff2ps -2eaz -w 8.5 -h 11 ${FAXFILE} | ps2pdf ${FAXFILE}.pdf) exten => in_fax,4,system(mime-construct --to ${EMAILADDR} --subject "Fax from ${CALLERIDNUM}

Hello I just read this article about a kid in England who built a box with a 3G SIM card: www.dailymail.co.uk/sciencetech/article-1394448/Doorbell-tricks-burglars-thinking-youre-home-invented-schoolboy-Laurence-Rook-13.html When someone rings your intercom, the box will call your cellphone so you can answer just like you were home. I don't know anything about electronics and would like to

Hello I don't have a landine and use a VOSP to provide access to the telephone network. In case a call comes in and I'm not home, I'd like Asterisk to log the call, and then send an SIP message to my VOSP so the call is forwarded to my cellphone and is thus charged to the caller, without Asterisk having to dial out to my cellphone through my VOSP at my expense and bridge the two

I have one analog line hooked in my Asterisk box using an x100p (I think that's the model number). When I do this in my extensions.conf: exten => 1200,1,playback(pls-wait-connect-call) exten => 1200,2,Dial(Zap/1/5555551212,20,rTt) exten => 1200,3,VoiceMail(u100@lightwavetech.com) exten => 1200,4,Goto,t|1 The phone rings beyond the 20 second timeout and never really goes to the *

Folks: I'd like to install Asterisk for use in my home. However, I'd like to continue using wireless phones in a couple of locations. The cheapest way to do this is to continue to use analog phone devices via an FXO/FXS box. However, I am not clear on whether I can expect these devices to provide "call waiting" features and caller ID features to the connected analog phone.