similar to: Monitoring calls through a transfer

Hello all, HI i am very new to asterisk and my boss needs me to investigate setting up asterisk for a new client. I have downloaded and installed (make, make install and make progdocs)asterisk on my personal computer and when i try to run it (./asterisk -vvvc) i get the following output below: NOTE: i am running REDHAT 9.0 on a 796MHz cpu machine: I am excited to be able to work with asterisk

Hi, I'm Andrew. (Hi Andrew) I'm a toll-free number junkie. I've had an account with iax.cc/sixtel for about a week, and every few days, I find myself sitting at the DID menu clicking the link that reads "Click here to get a random toll free number". I have three toll-free numbers now, and I don't know if it will stop... Is there any hope for me? -- Andrew

Stefan Gofferje wrote: > Hi Folks, > > on my home asterisk, I have a "huntgroup" for incoming calls on the > private line which first let ring my phones in my office and living > room, after a while then office, living room and bedroom. > I do this by simply putting two dial statements in sequence: > > > [private_huntgroup_day] > exten =>

Has anyone seen this before ? Feb 15 18:37:34 DEBUG[866]: That's odd... Got a response on a call we dont know about. I've got a whole load of them (328 in the last 5 minutes ...) Julian

Greetings, I'd like to thank everyone that has responded to my original email. I have received information from several companies, and will be testing several of them. I also would like to update a statement from my original message to clarify it: >My strikelist: nufone, voicepulse, iax/sixtel The strikelist is just a list of carriers that didn't meet the needs a resonable

Hi All, I have a server setup with an incomming PSTN line and a bunch of Grandstream BT100 phones. Is there a way for asterisk to foward an incomming callerID from the PSTN to the SIP phone that is setup as an extension? We have a Voice menu setup for incomming calls and I would like to recieve the caller ID of the calls we are recieving after the incomming caller reaches their final

Hi ALL; I have several users registered with my * box with different DID numbers, meanwhile they want to make international calls. When an incoming call comes in (over SIP), Is there any way that I can tell asterisk: 1) lookup among registered numbers 2) if not found in registrar , make an international call Appreciate any help MOhammad -------------- next part -------------- An HTML

Does any one know what happened with voice conduits? I have been trying to reach them for nearly three weeks now. Their voice mail boxes are full and writing email to them does not get any returns. Thoughts or sightings are appreciated. -- R.J.

OK, I'm trying to send an email to the list the contiune a thread which describes a problem I'm having. This particualy email I wish to send contains an ls -l describing my problem (too many open files) and is apparently too large to be considered a "normal" post, so I get a message that it's being held until a moderator can view it. Fine. So now I get an autoresponder

Hello, When I dial out there is a long delay in dialing. Is this normal? Thanks, David

I have a Quicknet Linejack in /dev/phone0. My phone.conf is: [interfaces] mode=dialtone format=slinear echocancel=medium context=mayores device => /dev/phone0 Only I can mark 7 digits, soon asterisk tries dial automatically. I cannot mark 8 or more digits. 6 or less digits work ok. -------------- next part -------------- An HTML attachment was

Hi I have a simple question but I cannot find the answer. I have a line with 2 different phone numbers I want to redirect each phone number called to a different IP phone Example Someone calls 5551234 and the call is redirected to IP phone 192.168.0.2 Someone calls 5551235 and the call is redirected to IP phone 192.168.0.3 Thanks Patrick

I would like to allow my remote users to dial in from their homes, cells, etc., and instruct Asterisk to forward calls made to their office extension to a number of their choosing. The wiki entry on "Asterisk call forwarding" shows how to do this. For security purposes, I would like to front-end this by asking the user to supply a password for their extension. Ideally, this would be

I was looking for something to connect a couple of POTS handsets to my asterisk server and found this on ebay http://cgi.ebay.co.uk/ws/eBayISAPI.dll?ViewItem&category=162&item=5162868118 &rd=1 The documentation says that it does SIP - therefore will it work in an asterisk environment. -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus

How long does it take to get a vanity number? I signed up for an account, pre-paid some money, and then placed a vanity number order. I did all of that around Dec. 31st 2004. They said it would take 2-10 business days. It is now Feb. 15th and still no vanity number. I've called them about a dozen times and every time they tell me to keep calling the number to check it and just wait. I'm

Has anyone had stability issues with IAX2. (Asterisk 1.0.5). reddwarf*CLI> iax2 show firmware Device Version Size iaxy 22 39344 I'm asking because in the last three weeks I've noticed the following two issues (on separate occasions): 1) Placed a phone call. Asterisk logs show the phone being answered and various files being Played back. But

Dear friends, Can i use the Asterisk functions (call recognition for example), using conventional telephony (in Brazil) ? Thanks in advace Pablo Fernandes

After updating to the latest CVS Feb 17 15:20:03 WARNING[15317]: config.c:819 read_config_maps: The 'sipfriends' table is obsolete, update your config to use sipusers and sippeers, though they can point to the same table. == Binding sipusers to mysql/asterisk/sip == Binding sippeers to mysql/asterisk/sip Feb 17 15:20:03 WARNING[15317]: config.c:823 read_config_maps: The

I'm using Dial to place a call to a PBX. But then I want to wait a few seconds and dial an extension. Dial doesn't return until the call is disconnected though. I also want the caller to not hear any audio until the DTMF has been sent. This gets the caller to the right place and he doesnt have to hear the welcome message from the PBX. Dial apparently isnt the application to use. Is

> > Having HZ differ between Xen and a guest doesn''t really matter that > much. The guest will get fewer upcalls than it expects, but it will > count ten ticks for each upcall. So I doubt that thsi is your problem. > Does ztdummy use the rtc driver? I''m not sure how weel we support that > on Xen... The problem is that the ztdummy driver assumes 1000HZ, and a