similar to: Grandstream wallmount??

I finally got the power supply for my 7960 and am having problems getting it working. What should be in sip.conf and the SIP(macaddr).cnf file? This is what I have in SIP0002FD3BA8F7.cnf # SIP Configuration Generic File # Line 1 appearance line1_name: Asterisk Test # Line 1 Registration Authentication line1_authname: "phone1" # Line 1 Registration Password line1_password:

I'm trying to get inbound calls from a TNT working but get 407 errors from the TNT. This is what I have in sip.conf: [maxtnt] type=friend host=x.x.x.x dtmfmode=rfc2833 callerid="MaxTNT" <maxtnt> context=demo qualify=yes disallow=all allow=g729 allow=ulaw insecure=very This is what the TNT is spitting out: Jul 24 14:55:12 tnt1 1/17: Releasing

I am using IAX to send a call to my cell phone. I want to be able to hit # and transfer it back into the office. I have added tTr to the dial command and hitting # prompts me for the transfer, but after I start dialing 103, it stops at 1 and tries to transfer it within nufone instead of my dialplan. This is the debug output: -- Called me@NuFone/1515480XXXX -- Call accepted by

Is there any interest in having ATM support for the various digium T1 cards? dave -- Dave Weis "I believe there are more instances of the abridgment djweis@sjdjweis.com of the freedom of the people by gradual and silent encroachments of those in power than by violent and sudden usurpations."- James Madison

Should I expect a standard fax machine connected to an ata-188 connected to an asterisk server, connected to a pri fed from a cisco 7206vxr to work correctly? It needs to have a standard fax machine, receiving and emailing it won't be acceptable. Thanks dave -- Dave Weis "I believe there are more instances of the abridgment djweis@sjdjweis.com of the freedom of the

I've never set up an mgcp device before. I have an Adtran IAD with the MGCP firmware on it. I have it configured in mgcp.conf like this: [general] port = 2427 bindaddr = 0.0.0.0 [adtran] host = 192.168.2.2 context = default canreinvite = no line => aaln/1 line => aaln/2 The device is configured like this: MGCP Configuration | Standard MGCP 0.1 / NCS 1.0 MGCP Endpoint

Should it work to have a multi-homed asterisk server with grandstream phones on the internal network and another grandstream phone on the internet and be able to call between them? I set the bindaddr to the external IP and pointed the internal and external grandstream phones to that address. The signalling works fine to call between phones, but when you pick up the ringing phone you get a

I am trying to configure cdr on a system. We are using nufone and I have set amaflags=billing on both of their sections in iax.conf. Incoming nufone calls show up in cdr with billing, but outgoing calls still show documentation. What do I need to change? We have a handful of SIP phones, 1 X100P outside line for local, and the rest is via nufone. I don't want inter-system calls to be

I've got a dedicated LD T1 terminating to a DMS100 switch. My outgoing calls aren't working, on the switch side they see two sets of dialing, with the first three digits repeating. I've used the sample extensions.conf modified a bit to remove the 9 and the 1, like so: [trunkld] exten => _NXXNXXXXXX,1,Monitor(wav,/tmp/dial) exten =>

Hello, I would like to know if voicetronix card (specially openswitch6 and 12) can be used with asterisk. Is there any driver for this card? Best regards, Daniel -- Daniel ANDRE (mailto:dandre@iris-tech.fr) IRIS Technologies - http://www.iris-tech.com Serveur kwartz - http://www.kwartz.com

Hello. I suspect the answer is no, but I'll ask anyway. Commercial phone systems have phones with multiple extension buttons e.g. 20, that can be programmed so that when you press one, it will call the extension. Is there any 'open' phone that can do this with asterisk, does an 'open' phone even exist. Further to that, commercial phone systems have a operators

Hi, I am new to Asterisk and wanted to ask a question concerning PSTN trunking. Is there a way to have DID's sent over IP to a switch? I know if One switch has traditional PSTN like a PRI this can be done, but is there a service provider offering this so I dont have to buy any tradtional PSTN trunking? Thanks, Jim -------------- next part -------------- An HTML attachment was

Does anyone out there use Asterisk with voip(sip or iax) long distance provider? Care to share about your experiances doing this? Michael

Hey all, I'm having problems reliably dialling out my FXO card. About 30% of the time I'll get a "your call cannot be completed as dialed". I'm thinking it might be the dialling speed, but I can't find any configs that change that setting. Any suggestions for troubleshooting? Thanks, Brad Waite

I am new to Asterisk and Digium card implementation issues. My VAR is strongly recommending using Apple hardware and Yellow Dog Linux for my telephony project, because of his familiarity with this OS. Is the PowerPC an appropriate and stable hardware platform for Digium/Asterisk development? Charles Hatchette chatchette@generalcare.com -------------- next part -------------- An HTML attachment

I'm curious as to folks experiences in selling asterisk-based solutions. In sales-speak, what are the common "compelling reasons to buy"? I can think of the following potential ones, but I'm keen to find out what seems to work in practise: - Customer wants to cut cost of calls, implements * and signs up to a VoIP/PSTN gateway - Customer wants a new PBX but doesn't want to

i wrote previous about a setup i thought might work with asterisk and the tsu-600. no one replied, so i thought i would ask if anyone is using a tsu-600 with asterisk and if so how do you have it setup ??

Hi all, Has anyone already written something which allows saving and loading the internal DB settings? All users CFWD and speeldial settings are stored in the DB in my setup which makes it a pain to restart Asterisk.... Looking at showtree in db.c (why isn't that exposed in the CLI?) It shouldn't be too difficult, but I don't want to reinvent the wheel. On the same track, I am also

Hi, I'm considering giving the Grandstream BudgeTone-102 phones a try. I've been using Cisco 7960's to date, but the low cost of the Grandstream phones are hard to ignore. I have two questions: 1) Does the message waiting indicator on the BudgeTone's work with Asterisk? 2) The one line 12-digit LDC concerns me a bit. Is the LCD able to display both the CID number and name on