similar to: AUSTEL Certified

Hello, I'm researching the possibility of using VOIP (SIP) with an existing PABX system. Ideally, the setup would be to dial an outside line through the PABX (that would actually link to the the VOIP gateway). At this point I would prefer not to purchase a hardware-based VOIP gateway. I would prefer to use a software-based gateway for research & testing purposes. Could anyone please

I'm looking to use Asterisk to replace my current PBX system. I'm in Australia so I need to use Austel approved equipment. My plan is roughly as follows: - Get a box with a suitable card and install Asterisk - Connect our existing PSTN lines to the Asterisk box - Get suitable softphones/ip phones to connect to the Asterisk system - Route all intra-office calls over our VPN - Route all

Semi-offtopic, Anyone know of voip providers who can provide tollfree number service? E.g. route 800-xxx numbers to our * ? Even better if they are familiar with * or can speak IAX ... -Dan

Hi, I'm new to Asterisk, and am researching information on linking Asterisk to an existing PBX. Could somebody please help me with what might be required for the following setup? - - We have an existing PBX. - I am going to setup Asterisk on our internal network along with some internal SIP phones. - I understand how Asterisk will act as the SIP Server, and SIP phones will be able to call

Hi, I am currently trying to find a replacement for a dinosaur PBX and want to replace it with a VoIP solution. We have just moved our lines over to an Optus Multiline from a Telstra ISDN Onramp 30 service with 100 lines. My question for you good people is what sort of hardware do I need to interface Asterix into the Optus Multiline? The Optus service is terminated in my office to a SHDSL

Hello, I have a T100P card on a Red Hat 8 machine with Asterisk installed on it and I'm trying to get the data routing properly through it. I purchased an Integrated Data/Voice T1 Line with channels 1-6 voice and 13-24 data. The line is plugged directly into the Asterisk machine's T100P card. Previous to the asterisk set up we had an Adtran router that connected everything to what the

Just a couple of questions for Aussie users/resellers! I have only just started to look at asterisk a couple of weeks ago and have found some very intresting discussions and some useful info on what can and can?t be done with it and technology. The questions i have is, Are there people using it is Aussie?( I would say yes so prob answered my own question ) :) Hardware, is the digium

Hi, I'd like to organize my Asterisk to properly handle incoming SIP/IAX/H323 callerids so they can be called back if needed. I have three incoming contexts for sip, iax and h323 calls. To each incoming call I'd like to prepend certain number that will be catched with pattern matching on output calls. For instance for iax I have: [from-iax] exten => s,1,NoOp(IAX call from outside

Hi, some questions/comments about performance/utility of * and * hardware I've been reading this list for a few weeks and I think I have compiled the better feelings of the users. please correct me if I'm wrong, still learning * .... Will be nice to see something like this in a wiki. After being flamed and corrected I will repost "clean" data. 1- Transcoding is the process of

Hello all, I was wondering if anybody knows where one might obtain a PCI ISDN card supporting a single BRI for use with Asterisk in Australia (and using something like chan_capi). Because of the Isdn4Linux DTMF issue, I don't want one of those cards. I've already spent too much time messing about with my current card. I'm after something like the AVM Fritz! cards. I found one place

Hi, Last weekend I was planning to buy a physical PBX system, but instead I have been blown away by the fact that VoIP really works, that Asterisk is so easy to set up and use... and free! We're in Australia, so as I understand it, we aren't allowed to use the Zaptel cards. We need to set up our system to route incoming 56K data calls to the PPP daemon on our Linux box or to our

Four years ago, I faced a real dilemma in my business: the Visual Voice PRI dll had a bug that considered unanswered any call after ringing for 20 seconds. This bug was in fact killing my business, because for international calling, the setup of the call was already close to 20 seconds on many cases. Furthermore, the vendor, Artisoft, had cowardly sold the software to Dialogic, and Intel-Dialogic

We are not having any luck with the E100p card here in Australia, it will work with a crossover cable to another device but will not talk to our Telco Telstra who probably have a weird implementation of an E1. Any suggestions on a replacement? Regards Mark McKibbin DCS Internet 64 Queen St Warragul Victoria 3820 Australia www.dcsi.net.au mark@team.dcsi.net.au Ph. 1300 665575 Fx. 1300 556595

I just had a user complain about a call getting dropped and another one failing to go through. I'm trying to interpret the log entries for each call and would like to confirm my understanding. The first entry is from an outbound call to a 1-866 number. The 5058971234 at 216.31.xx.x refers to a user account on my server. Voip_one refers to my provider. This sequence of log entries

I have a linejack and a phone jack in my asterisk server working well between the SIP phones and the phonejack. what I cannot get to work is the outbound linejack Phone/phone0 trunk line? how can I get a SIP or Phone/phone1 phonejack phone to dial 9 then outside number and pickup Phone/phone0 and dial it? right now it accepts a 95551212 but busy's on the last digit 2. no outside dial.

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Hello, Does anybody have the latest Boot ROMs for the IP500 and IP 600 Polycom phones. I have one of each and can't find the Boot ROM v 3 anywhere to download. I would also love a good sample phone.cfg and sip.cfg files from an Aussie asterisk user to look at. Also the ip500 is having problems trying to load the bootrom 2.6.2 ? Any ideas? Kind regards Michael Felder IT Medic

Hi All, I am using a Sipura SPA-3000 as an FXO gateway to bring calls in and out of Asterisk. I am using "PSTN Ring Thru Line 1" (on the "PSTN Line" tab) so Asterisk answers the call rather than the SPA-3000. It is all working perfectly except I can't get the SPA-3000 to pass caller ID to Asterisk. It passes "Display Name", "User ID" and any "PSTN

Hello all, Been using Samba for a looooonnnng time, and with great success I might add. This is the first time in EVER that I've needed to ask ANYTHING regarding it, so easy is it to config and admin. Kudos to the Aussie that wrote it. Even if he initially thought it was Pathworks... 8-) ANYWAY... I've never really been able to use Samba as a print server. For some reason, when

Hi I have set up dovecot with a self signed SSL certificate, created with the mkcert.sh script. Now I see the following in the log: pop3-login: SSL_accept() failed: error:14094416:SSL routines:SSL3_READ_BYTES:sslv3 alert certificate unknown [206.124.112.22] pop3-login: Disconnected: rip=206.124.112.22, lip=204.3.153.71, TLS handshake Is this caused by my self certified certificates. Or is it