similar to: Some advice on routing DID's

Kevin, Are you in the US? If so then you've probably got several carriers to choose from. In my experience analog lines have a flat expense of $20-$25 per month. That equates to about $140-$175 per month in flat fees, plus you have usage on top of that. (Your experience may vary.) I am currently experimenting with a company out of NY called Digizip (www.digizip.com) that sold me a Qwest

http://grandstream.com/BETATEST/HT488_496_386/

I am looking to setup paging using the auto answer feature on the Grandstream GXP2000. I am thinking I will follow the method as described here: http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+page I will setup the 4th account on the phone to auto answer. Does anyone else have a method that works better? I also looked at the allpage AGI written on Voip-Info. But it seems

I see that Asterisk@home and FreePBX are going along similar lines with web based management interfaces.. My Asterisk box has one analog phone, one analog line, 3 SIP phones, IAX inbound numbers, an IAX outbound trunk, IVR menus and voicemail boxes in different contexts for each of the inbound numbers.. Soon I will be adding one or more IAXy devices.. Would either Asterisk@home's or

Hello, I would like apply different call rate (tarif) per outgoing number (or group of phones, based on prefixes), I'm playing with astpp, but seems, that this feature isn't available here, can you recommend any other open-source billing (A2billing, AstBill?), that this can do? thank you! PJ

Not sure what's up with the mailing list here. For some reason mails are not coming through. Try again... I am trying to link an asterisk box to my provider's asterisk server via SIP. (I know I could use IAX, but the provider does not allow that, so I can't). When an inbound call happens I get this: Jan 31 13:09:14 NOTICE[3716] chan_sip.c: Failed to authenticate user

Does anyone have experience setting up ASTPP? I have an Asterisk server in my office that I also give access to some friends and family that live outside Mexico so they can make local calls. I want to keep track of the costs and I only need to use ASTPP to rate the calls, not for calling cards or anything else. I found the documentation a little vague on the details so after setting up and

I do part time consulting work. I need to setup an asterisk system to allow me to record both inbound and outbound calls to clients. I have one client that is just a PITA. The client has changed their mind three times so far and we are at step one. I have a spare slackware box and a seperate phone line for the consulting work. I have MCI Neighorhood as my carrier. What I need to know is: 1.

Hi to All, I've an Asterisk CVS Head working with Mysql. My problem is that instead of ANSWERED or something like, into the CDR database records, I find only numbers. This is also a problem to let ASTPP works, infact I receive an error: ERROR - ERROR - ERROR - ERROR - ERROR DISPOSITION NOT MATCHED and the call has no cost. Any suggestions? Thanks -- .:FaberK:.

Hello All, I am having trouble finding a specific * piece of software so I thought I would see If you guys can help me get my terminology clear. First off let me premise this with "no, this is absolutely not for doing call marketing". I need to make my Asterisk box call a group of people and play them a message. My company deals with education so we need to do follow ups if students

Hi all, At our customer site i've installed one asterisk server with 20 Grandstream GXP2000's. Firmware 1.0.1.9. When someone dials the customer, the receptionist picks up, and does an attended transfer (the 'grandstream way') to a collegue. Most of the times this goes ok, but sometimes, when the receptionist puts the call on hold, and tries te reconnect to the caller there's

Does anyone have a mysql query that will compare a number from the asterisk cdr to a table of international country+city codes to determine the closest match? The two fields are; 1. Asterisk mysql cdr 'dst' field - sample record value '011441316551212' 2. rate table data like this DialPattern 011447977 011447979 011447980 011447981 011447984 011447985 011447986

Having trouble running make on res_perl: [root@charlie res_perl]# make perl -MExtUtils::Embed -e xsinit gcc -Wall -fPIC -DHAVE_AST_CUST_CONFIG -I/usr/src/asterisk-1.0.9/ -I/usr/src/asterisk-1.0.9//include -I. -DAST_INSTALL_PREFIX=\"\" -DASTETCDIR=\"/etc/asterisk\" -DASTLIBDIR=\"/usr/lib/asterisk \"-DASTVARLIBDIR=\"/var/lib/asterisk\"

Hi, What do I have to do to get local\number to work in a context? It works from my [from-internal]... however from subcontexts it does not work: Jan 6 15:55:32 VERBOSE[20237] logger.c: -- AGI Script Executing Application: (Dial) Options: (Local/570323xxxx) Jan 6 15:55:32 NOTICE[20237] chan_local.c: No such extension/context 570323xxxx@default creating local channel Jan 6 15:55:32

Good Day, I'm finally getting around to officially announcing ASTPP. For the last 6 months, I've been working on converting ASTCC into a decent billing package for asterisk. I'm just finishing up fixing a few bugs before the 1.0 release and would appreciate if there would be a few who would be willing to do some testing on the software. Here is a list of features: Provide call

Hi All, I'm looking for a good voip hardphone that has a decent set of the "regular" features (conference, 2 lines, etc) thats reliable, has decent quality, and isn't too pricey. Does anyone have any suggestions? Thanks in advance. Derek -- Derek Fedel

Figuring I knew what I was doing (I didn't - surprise) I added a totally unnecessary line in /etc/asterisk/extensions_additional.conf a couple of days ago. Troubleshooting a dialing rule issue, I'm now realizing that FreePBX is updating its database with the new settings but is not rewriting/updating extensions_additional.conf with the changes I'm making. I've tried renaming the

so what should "astpp db" be exactly, where can i find its name? what should i write there? Thanks again.. > The Database field should contain the name of the astpp db, something > along the lines of "astpp" is what I would put in there. Here is a fixed > version of the script. It did not post properly to the wiki: >

I am getting a TON of these errors on the console. I Googled and wikied and greped found the error in the source but cannot understand why it is happening. The system works fine, no dropped calls, no echo, it will even run for weeks with this error. But it just scrolls and scrolls on the console. Temporary fix was to turn off the console monitor! :-) Any ideas. Apr 16 10:40:12

Anyone have any luck RMAing a Sipura phone since the Cisco take over? Sipura only has support via email or fax to end users and I haven't gotten a response to either for over 2 months. Linksys Support will jump you through all their scripted hoops to resolve your problem (they hope if they speak with a thick enough accent and make repeat the same steps over and over again that you will just