similar to: IAX2 authentication confusion

Hi, My question is in refernece to the posting by Jeremy McNamara here.. http://lists.digium.com/pipermail/asterisk-users/2003-October/022966.html He states that in order for "trunking" to work the type has to be peer.. When I set mine up I did so using type=friend just to make it simple.. So when I read the above posting I thought well maybe my "trunking" has not been

I am trying to get two * boxes to communicate with eachother. I have read http://www.voip-info.org/wiki-Asterisk+-+dual+servers as well as information on IAX channels, the Dial() command, and the switch statement in extensions.conf. But I am having no luck. I have a working * box running with a Zap card that I want to be the server machine. I have another little box running * with just a

On 03/11/03 00:25, Mark Spencer wrote: > As a side note, I strongly would like to see someone implement a > client using libiax2 which implements IAX2 instead of the (now > obsolescent) IAX version 1. I'm implementing a Java-based IVR server (and yes, I know Asterisk does IVR, and no, it's not flexible enough to do what I want and no, it doesn't integrate well with the Java

Im having problems setting up a trunk between two locations. Heres the setup I have: Server A is connected to the PSTN at my datacenter Server B is connected to a clients e1 line at his datacenter I only want to route calls from Server B to Server A and out through the PSTN. Server A has a lot of other things connecting to it, so I need a very specific context for all calls to go through.

I've turned on debug in a (IAXComm based) soft phone. I see the phone sending pings to *. I see * getting the pings. For some reason, with iax2 show debug, I never see any response on the console from *. However, the phone shows a response with INVAL. Seems like an odd response to a ping request. I believed it should get an ACK. Is that wrong? If I should get an ACK, what could I have messed

http://www.voip-info.org/tiki-index.php?page=Asterisk+IAX+authentication After struggling to understand IAX2 inbound authentication, I figured I'd document what I learned so others can benefit :-) All comments welcome, especially if I misunderstood or improperly explained anything.

Dear all i'm trying to originate an outgoing call with the command originate, from Asterisk's CLI i'm typing: CLI> originate IAX2/my-iax-provider/number2call application wait 10 [Apr 18 19:31:12] DEBUG[32331]: chan_iax2.c:4000 create_addr: prepending 40 to prefs -- Call accepted by 62.149.202.150 (format ilbc) -- Format for call is ilbc -- Hungup

Hi, The iax.conf is below and the trace. Any ideas please? disallow=all allow=g729 trunk=yes qualify=yes qualifysmoothing=yes nat=yes canreinvite=yes context=OutboundWS transfer=mediaonly Executing [082449627 at private:1] Dial("SIP/919-094d6e60", "IAX2/ECom-iax/2782449627|60|") in new stack -- Called ECom-iax/2782449627 -- Call accepted by xxx.xxx.xxx.x (format

I'm getting exactly the same behavior as was posted about in http://lists.digium.com/pipermail/asterisk-users/2004-March/040380.html I've upgraded (both ends) to CVS stable (CVS-v1-0-06/17/05-13:15:49). Jun 17 13:46:17 NOTICE[15942]: chan_iax2.c:4053 authenticate: No way to send secret to peer 'a.b.c.d' (their methods: 4) Immediately after that, I'll see frames go by with

Greetings folks; I guess I must be missing something, because for the life of me I can't seem to make this work. I have remote clients connecting to Asterisk using IAX2, these clients have changing IPs so we're using the useful register tool. The client can call out successfully, that's not an issue at all. Calling coming from the server to the client, however, do not appear to go

Hi, I'm slowly getting to grips with *. My next quest is to get IAX2/FWD calls working. I've setup a fwd account and added IAX capability to it via the website. I got the email saying it had been done with some welcome text and sample phone numbers to try, such as 10001 for the answer phone. I followed the setup example on the fwd site for configuring * to work with fwd's IAX.

Hi all, I'm using following software and equipment and I have very strange behavior: Asterisk CVS-NHEAD-05/30/05-16:42:41 H323 gatekeeper - GnuGK 2.2.2 IAX2 client - Firefly 1.9.8 build 3945 H323 client - SJPhone Build 1.50.271d H323 gateway - Welltech Wellgate 3504A When I dial from Firefly (IAX2) -> SJPhone (H323) everything works as expected. When I dial from SJPhone (H323) ->

Hi, I have some questions regarding IAX, IAX2 and encrypted authentication. How can I know if IAX or IAX2 is used between two * servers? There is any guide about how to configure encrypted authentication (not in clear text)between two * servers? I "hear" on this list a couple of days ago that port 5036 is the default one for IAX and something else (4XXX) for IAX2. Trying 'iax

I am trying to connect an IAX ATA to an Asterisk 1.4.21.2 system. The Asterisk system has been stable for years, and has no trouble bridge SIP phone sets to IAX trunks. When I initiate a call from the IAX ATA, something goes wrong. One rare occasion it works fine, but usually there is no audio passed. I have a snippet of the console below. Notice no bridging message...not sure if that's

Is nufone having problems taking gsm calls today i had some issues dialing overseas to call my folks. here's snip of what the console displayed -- Executing Dial("SIP/2204-a279", "IAX2/fgravato@nufone/011351217907000|100|T") in new stack Jan 18 10:30:02 WARNING[1200884528]: chan_iax2.c:5036 iax2_request: Unable to create translator path for UNKN to GSM on IAX2[NuFone]/1

Hi, I have two asterisk machines, chomper and otao. otao is otao.ieworks.net, has a public IP address (66.101.11.61), but no PSTN connections. chomper is at my house, behind NAT, but has a single X100P PSTN connection. I would like to establish two way calling between otao and chomper. Right now, I can call my extension on otao (2101) from my x-lite softphone on chomper, but I cannot call

Hi, I have two asterisk machines, chomper and otao. otao is otao.ieworks.net, has a public IP address (66.101.11.61), but no PSTN connections. chomper is at my house, connected to cable modem behind NAT, but has a single X100P PSTN connection. I would like to establish two way calling between otao and chomper. Right now, I can call my extension on otao (2101) from my x-lite softphone on

I've asked this before but I don't recall seeing anyone respond. Maybe the answer is so simple that it should jump right out, but its not. I've got three iax connections defined, two working fine (including one to iaxtel), and one that is somewhat intermitant. If I do a cli 'iax show registry' and 'iax2 show registry' I see "both" iax and iax2

Hi My head hurts... Can anyone help out here, my remote IAX can see my local IAX and visa versa, conversation starts, I can dial my remote (POTS) landline number, remote end answers, trys to route to local iax2, I see it start the conversation here, the extension (SIP) rings once and then it dies... Both ends are defined with accept IPADDRESS to keep it in the family and simple.. Debug info

Can any one help? Thanks. we have two * servers (Version 1.2.1) and one 1.09 server. Calls between these two 1.2.1 servers have odd behavior. But call from 1.09 to 1.21 server working fine in either situations. See below pls: Local server iax.conf [tosyd] username=mel type=peer secret=xxxx host=198.168.2.66 remote server iax.conf [mel] type=user secret=xxxx host=198.168.2.67