similar to: 1.4 and IAX Trunks ...

I've setup this simple configuration to test the new mediaonly iax feature in 1.4 : Input (client) -> Server (routing) -> Termination transfer=no transfer=mediaonly transfer=no all the machines are in the same 192.168.0.x net the routing Server in the middle has iaxusers realtime backend on mysql the call is originated with a sip phone registered on the Input client

I have two asterisk servers connected using IAX. Server A has a TE410P running on a Xeon 2.4Ghz with 2GB RAM and 36G IDE HD on Debian 2.6.11-1-686 and Asterisk CVS-Nv1-0-7-06/01/05-01:27:25. Server B does not have any Digium board, but has ztdummy and zaptel loaded. It's runnin on a P4 1.6Ghz with 1GB RAM and 36G SCSI RAID 10 on Gentoo 2.6.11-gentoo-r9 and Asterisk 1.0.7. The

Hi, I posted this on the Asterisk forum months back with no real answer() so i'll try here :o) Details: There is 3 asterisk boxes called X, Y and Z.. all boxes peer with each other via IAX2 and have dialplans setup... etc etc There will be asterisk based clients connecting via IAX2, and for example i'll call them A, B and C The clients only peer directly with one of the X, Y or Z

Hello! I have a number of IAX clients behind a NAT (on the same LAN) and asterisk server on the Internet. And that clients doesn't speak directly to each other, traffic goes through the asterisk server. What should I configure to make IAX clients on the same LAN to speak directly, please? notraster=no is set in iax.conf The asterisk server is on real IP behind a NAT (at DMZ with full 1-to-1

Hi all, i have the following setup PSTN -> routing server (asterisk 1.6.2.11) -> IAX -> callcenter asterisk 1.6.2.9 -> SIP -> agent Does work quit fine - then agent does have the abibility to transfer a call to a third party - the agent can initiate the transfer over a web interface - it does generate a asterisk manager atxfer request... So agent does initiate transfer - call

Two asterisk servers, one running a recent HEAD, the other 1.0.9. I have both ends set up with trunk=yes, notransfer=yes, type=friend. I notice that the trunking works from HEAD to 1.0.9 only (the direction in which calls are originated). I know this by bandwidth usage and by iax2 trunk debug. I did have to use trunktimestamps=no on the HEAD end to keep it quiet. I assume this is the new

I have a situation where our VOIP provider is running *, my office is running *, and my house is running *. I have an extension at the office so that if a call comes in from the VOIP provider and they select that extension, the call will be sent to my home * box and ring my phone. That works fine. I set "notransfer=no" in the iax.conf file at the office so that the office system can

Hi, My problem is related to a bug in the Asterisk bug database: (Bug 2773) http://bugs.digium.com/view.php?id=2773 Essentially, when one has a call going over more than one IAX legs, the audio is not transferred *sometimes*. This is quite randomly observed. With "notransfer=yes", the problem goes away. In my situation, the IAX channels originate from 2 Asterisk servers themselves -

On Sat, 2009-02-07 at 15:51 -0500, Alexander Lopez wrote: > > > -----Original Message----- > > From: Steve Murphy [mailto:murf at digium.com] > > Sent: Saturday, February 07, 2009 1:59 PM > > To: Alexander Lopez > > Subject: RE: [asterisk-dev] 1.4 and CDRs -- The Breaking Point > > > > On Fri, 2009-02-06 at 12:28 -0500, Alexander Lopez wrote: >

On 11.9.0: > -- Accepting AUTHENTICATED call from 111.xxx.yyy.zzz: > -- > requested format = speex, > -- > requested prefs = (), > -- > actual format = ulaw, > -- > host prefs = (silk16|ulaw|gsm|g722), > -- > priority = mine > -- Executing [8447 at voip-in:1] Dial("IAX2/n4-5734",

Hello, Right now there is not a table build script at: http://www.voip-info.org/wiki-Asterisk+RealTime+IAX Therefore I have taken the SIP build script and added a few fields that I use from my iax.conf (could be more out there, please see the complete build script below): `dbsecret` varchar(100) default '', `notransfer` varchar(100) default '', `inkeys` varchar(100)

We have a problem with our iaxclients. Our asterisk runs on a public host with debian and many of our IAX2 clients are natted. The iax.conf looks like: [23456] accountcode=123 type=friend context=user auth=md5 secret=xxxx username=23456 callerid=Testuser 1 <23456> notransfer=yes host=dynamic The cli command IAX2 show peers shows all clients as unmonitored CLI> IAX2

I have created an asterisk.org blog entry: http://www.asterisk.org/node/48358 to describe what I will shortly be committing to trunk to correct the weaknesses of CDRs, that asterisk users and developers have been complaining about for quite some time. Highlights: Restructuring the code and philosophy of CDRs. Plans to eliminate the ForkCDR() application Plans to create

Hi there The default option for saving graphics from R (1.9.1) on my Mac is as a pdf file. If I open the file in Acrobat reader it looks really good and crisp, and is obviously saved as vector graphics, since I can zoom in as much as I like and it continues to look really nice. If I import it into MS Word (from office 2000), or Textedit, however, it imports it as a bitmap and unless I save

hi all, I'm trying to set up some iax2 trunks in Realtime architecture with the same backend. All work better (make call, receive etc etc) but when I do "iax2 show peers" some asterisk don't show anything and other show the iax2 peers but with status "unknow". Name/Username Host Mask Port Status ctm1/trixbox 10.0.0.131 (S)

Let's say i have a network of * boxes connected via IAX, one of them is a "switch", one or more are the "gateways". - An IAX[2] "customer" register himself on the "switch" (and gets an accountcode for te purpose of cdr) - The customer places a call to the "switch", the switch does some magic and decides which "gateway" the call

Hi All - Well, after happily existing in a one office environment with asterisk for a few months, I've now decided to start adding in our other offices with their own * boxes and IAX connections (over VPN). Unfortunately, I'm an idiot and I can't get it to work. I'm having some kind of problem with codecs, I guess, but I don't understand what or why. When trying to use

Trixbox 2.2 Beta with freePBX 2.2.0rc1 I have a setup that looks something like this in ASCII art: Teliax IAX Trunk ------+ | V Embarq PRI ----> Tandem switch ----> Ottawa Office Server------+ +--------------> Lima Office Server -----+|

I am trying to do the following: 1. Call comes in to my * box over IAX (VP Connect DID) 2. Check to see if call should be forwarded to my cell 3. Forward the call to my cell phone and take * out of the media path. I am able to do all of the above except * is not able to natively bridge the call. I am using sixtel and for the call forward portion, but the calls don't connect before sixtel

-- Attempting native bridge of IAX2/peer1-iax-7 and IAX2/peer2-21 I want everything to stay in the VoIP server rather then briding. I have notransfer=yes on, but it still seems to bridge the call natively.. can I keep the RTP stream on the asterisk server some how?