similar to: asterisk and pingtel

This is not really a problem just something I noticed in my testing.. When two or more Asterisk servers are connected by IAX2 trunks it does not make use of any "shortest path" type system.. (maybe this is still planned somwhere down the line, but may come in handy to those who have multi asterisk installations) Here is the setup.. UA1--- Asterisk1----[IAX2 Trunk]---Asterisk2---UA2

hi, i saw the cdr file called Master.csv and i want to know what these represent. examples "","","4","incoming","","Zap/1-1","Zap/4-1","Voicemail","u8888","2003-11-07 17:43:04","2003-11-07 17:43:04","2003-11-07 17:43:22","ANSWERED","DOCUMENTATION"

asterisk*CLI> load cdr_unixodbc.so Loaded /usr/lib/asterisk/modules/cdr_unixodbc.so => (unixODBC CDR Backend) == Parsing '/etc/asterisk/cdr_unixodbc.conf': == Parsing '/etc/asterisk/cdr_unixodbc.conf': Found -- cdr_unixodbc: dsn is MySQL-asterisk -- cdr_unixodbc: username is root -- cdr_unixodbc: password is [secret] -- Connected to MySQL-asterisk it

I would just like to get a refresh of the situation for Australian users. It would seem that the TE400P is currently available and is likely to acheive approval for use in Australia within the next 2 months ? (when is the end of summer?). Once this is done, it will certainly suit larger installations, but it still leaves a number of 'gaps'. Anyone with analog phone lines will need

In the absence of "The Definitive Guide to Asterisk Dial Plans" book, I'd like to do something possibly unique with the formatting of extensions in my dial plan, and am having trouble. We use VoicePulse connect, which gives us local DID for inbound and outbound calls (even though DTMF tones are not working in Voice Pulse Connect at the moment). To dial local numbers, you have to

Does anybody have useful tips on creating good quality recordings for use with prompts in asterisk? I'm interested in hearing input on hardware (mics, dats, sound cards, etc) and software (recording software, dsp) as well as recording techniques. Jerimiah Tularosa Communications

Good Morning, I'm working with a queue at the moment and I having trouble with my digits. Australia is my example. On the tiki it says for international digits, I can dump them in the "digits/au" directory. I tried that -- just because, I also made a copy in "au/digits". When the queue announces the position I it says: -- Started music on hold, class

Slash-dot is pointing to this article on Asterisk and Pingtel. http://www.theregister.co.uk/2005/05/22/pingtel_voip/ Paul Paul Mahler www.signate.com

Hello all, I have setup my sip.conf so users can register etc in the following format, [person] type=friend username=nick secret=******** host=dynamic mailbox=101 in my voicemail.conf I have an entry like 101 => 1234,Nick Knight,nick@omniis.com Leaving a voicemail works fine after I have my dial command time out but on sip clients which display whether voicemail is

I've seen this question asked before but haven't seen a definative answer. Does the e100p work in australia? Did any one who was asking the question before bite the bullet and get one? I can get a te410 if i really have to but would prefer to stay with the cheaper option. Any comments appreciated, Thanks ------------------------------------------------- This mail sent through IMP:

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

Hi list! I have a little question about groups and Asterisk ... is there anyone out there that can say if Asterisk can do any of this; We have a customer that want call handling we cant give him with a traditional PBX, and I'm running Asterisk @home so I thought I could give it a try ... The customer wants that incoming call should go to one group with some phones in it, if the group is

In the past two days, I've been experimenting with the Pingtel SIP softphones and Asterisk, which one of my customers has been using. A few notes: 1) The "insecure=1" setting in SIP peers now works, from my limited experiments. Mark put this in for SIP servers that don't send requests in with a return port of 5060. This flag essentially takes any request inbound _to_ port

Hello all, Hope I am not too of topic here - but it cross's the phone/asterisk boundary. I have been playing with a few soft phones - noticed that pingtel seemed to be highly recommended across previous postings. I have been using xten - which is a great phone but seems a bit limited in its functionality - which is why I am now looking at pingtel. Problem is I cannot get it to

Has anyone been able to get conference calls to work on the Pingtel Phones? I assume this feature works with their implementation, but connected to my asterisk box it doesn't work. The Pingtel phone thinks it is making a second call, but asterisk never sees anything about a second call. Any help would be appreciated. Sincerely, Andy Hester Consero

I just bought a Pingtel Xpressa from VoipSupply for use with Asterisk. I know that Pingtel has sold off their hardphone line and discontinued support for their phones, but I'd like to track down a few of the Java applications that they distributed before they went away, specifically their LDAP Phonebook app. Does anyone have a copy that they could send me? It was publicly

If you can in any way improve this page: http://voip-info.org/tiki-index.php?page=Pingtel+Hardphone please do. Thanks very much. Jason Sjobeck -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050717/d9b379e1/attachment.htm

1) Who bought Pingtel's phone line? 2) Anyone seen this chinese-made VoIP phone that supports 8 different protocols? http://www.telecom.globalsources.com/GeneralManager?language=en&design=c lean&action=GetArticle&article_id=9000000055338&page=printarticle&printT his=yes Mike :) -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi Vic, I hit that same problem! My SIP phones would sound okay when I made changes to indications.conf but incoming calls in to my TE410P had their own thing going on! Have a look at the zaptel source files, there's one called zonedata.c. You'll see the au settings... replace what's there with this: { 1, "au", "Australia", { 400, 200, 400, 2000 }, { {

First I like to apologize if this is common knowledge, but I'm unable to get MSN messenger 4.6 to register with asterisk. I configured MSN messenger to use UDP and the IP of my asterisk server I edited the registry entry - for pC2PC calls under Windows98. What I'm I missing ? Asterisk version information Asterisk CVS-04/25/03-05:37:19 sip.conf [pingtel] type=friend