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This page http://www.freecall.com/en/index.html is advertising free calls to: Argentina, Australia, Austria, Belgium, Canada, Czech Republic, Denmark, France, Germany, Hong Kong (+mobile), Hungary, Ireland, Italy, Luxembourg, Malaysia, Monaco, Netherlands, New Zealand, Norway, Panama, Poland, Portugal, Puerto Rico (+mobile),...

...ster to talk to each other. Calls withing voipblaster network is free. If I configure two sip clients?with the two accounts it works fine however with Asterisk I am getting SIP 401 ? In my Sip.conf file I?under general ? register = user:password at sip.voipblaster.com ? then I have a sip peer ? ? [FreeCall](default) type= friend context= incoming username = kiks2010 secret = password host= sip.voipblast.com fromuser = kiks2010 fromdomain = sip.voipblast.com insecure=very qualify=yes ? these are the sip debug logs ? v=0 o=kiks2010 1287592622 1287592622 IN IP4 77.72.168.99 s=SIP Call c=IN IP4 77.72.16...

...the user option, what have i missed? I'm using samba 2.0.7 on RedHat 7, help would be much appreciated. -- --------------------------------------------------------------------- Chris Jensen cjensen@edex.com.au Educational Experience (Australia) Postal Address: PO Box 860, Newcastle NSW 2300 Freecall: 1-800-025 270 International: +61-2-4923 8222 Fax: (02) 4942 1991 International: +61-2-4942 1991 Visit our online Toy store! http://www.toysandmore.com.au/ ---------------------------------------------------------------------

Hello, I have a 4 port FXO digium card with 3 PSTNs attached to it and AsteriskAtHome setup. Everything is working fine except outbound calls. When I dial a outside number, it works fine, but when another employee trys to dial out while I am on a line, it will not go. I have a outgoing route setup in the AMP interface. Dial Pattern: 1NXXNXXXXXX NXXNXXXXXX NXXXXXX Trunk

Hi, I am having a few dial plan problems which I wondered if anyone would be able to help with. Firstly, I wanted to send 0800 calls through 1 sip provider and other 08xx calls through another. I have this: exten => _0800.,1,Dial(SIP/${EXTEN}@provider1,30) exten => _0800.,2,Congestion exten => _08.,1,Dial(SIP/${EXTEN}@provider2,30) exten => _08.,2,Congestion However,

...t the following on the console Mar 21 05:54:15 NOTICE[68071]: chan_iax2.c:6123 socket_read: Rejected connect at tempt from 203.13.163.245, who was trying to reach 's@' the s part i can understand by the @nothing ......?!? my iax.conf looks like this register => aaaaaa:bbbbbbb@proxy.freecall.net.au [alphanet] type=friend username=aaaaaaaa auth=plaintext ; ugh plaintext secret=bbbbbbbb host=proxy.freecall.net.au context=main disallow=all allow=ilbc my extensions.conf looks like this [default] exten => s,1,Answer exten => s,2,Dial(SIP/me,40,tr) also I have the same under [main...

...ar [gs486] type=friend context=default username=user secret=pass host=dynamic ,nat=no canreinvite=yes dtmfmode=info call-limit=2 allow=all [budgetphone-out] type=friend context=inbound secret=pass username=user host=budgetphone.nl fromuser=user fromdomain=budgetphone.nl insecure=very allow=alaw [freecall-out] type=peer username=user secret=pass qualify=no authuser=user fromuser=user fromdomain=freecall.com host=sip.voiparound.com insecure=very canreinvite=no allow=all [iXcall-out] type=friend context=inbound secret=pass username=user host=ixcall.net fromuser=user fromdomain=ixcall.net insecure=ve...

I am having problems getting incoming caller id to work on a Telstra Onramp 10. I have changed "/DEFAULT_CIDRINGS 2"/ Is there something i'm missing ? My Cisco 7960 just shows "asterisk" Thanks, Nathan [zapata.conf] context=incoming usecallingpres=yes relaxdtmf=no rxgain=0.0 txgain=0.0 busydetect=no pridialplan=local usecallerid=yes callerid=asreceived

Hi all, I've been through the wiki and the archives and I've been unable to find what I'm looking for. Basically I have a phone that I don't want to be able to dial out. I've only got one context at the moment; but from my investigations I think I might need to create another for this specific phone. Can anyone point me in the right direction? Thanks Matt

...go ahead to start dipping its foot into the shallow end of the asterisk pool. The client we will be setting up for currently has an NEC PBX of some sort with 8 analogue lines in. They use lines 1-4 as indial on a rotary group, lines 5-6 as indial for two 1800- numbers (1800 is Australian prefix for freecall), and lines 7-8 as outgoing. There are approx 15 extensions using some sort of non-standard handset. During the quiet period for this client (the rest of January) we will be configuring asterisk on a new E1 service with a few analogue and a few VoIP extensions. The current analogue lines to the N...

I'm wondering if anyone out there is running a community PBX for their local Asterisk User Groups or area Linux groups. I've been thinking of setting one up but am stuck as to what services to provide that people would actually find useful. I know that I could setup simple SIP->SIP to allow everyone to call each other, but that's not generally too fun. -- Kyle Sexton

i am running MsOffice 2000 in wine-20031212 without problem. Now i tried to do the same with wine-20040408 but its not working. word stops with showing lot of errors and last with "winedbg>" Any body running office 2000 in this version. Please mail ur config file or any tweak i want to work on it. Thanks SK --