similar to: IAX2 calls failing one way.

I received this before, and it is because you are using the wrong context in the iax.conf. For example: The context must match the username in the register statement. Iax.conf... register => username:secret@iax.host.net [username] type=friend context=iax-in user=username secret=secret auth=plaintext host=iax.hust.net ----------------------------------------------------------------

I had a working 1.0.9 asterisk installation and tried to get a Firefly IAX phone to register, but it was failing. I upgraded to asterisk 1.2.5 and the PBX is working fine, but the IAX phone still won't connect. Below is my iax.conf and the output from setting iax2 debug while the phone tries to connect. Could somebody please give me some pointers? This doesn't seem to be a normal

For some reason, starting just today, 1 out 3 of my asterisk servers is having issues calling 1 other server. The only issue I see is that when it registers with the problem server it is using port 1027, not 4569. ie: Registered to 'Server 1', who sees us as 'Server 2':1027 Server 1 then proceeds to timeout trying to register with Server 2. The way I have each server registering

Hi All, I'm having problem with IAX, I'm trying to connect to speex.co.il from asterisk using: register => username:password@speex.dyndns.org and I cant get it to work. Maybe someone who already got this to work will help... When dialing my speex extension I see the next output from consol: IAX2 Debugging Enabled *CLI> Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno:

Hallo Did you allow udp outgoing on 4569 as well.. i found udp bit different than tcp when comming to firewalls liaan ----- Original Message ----- From: "Bartosz Wegrzyn - asterisk" <junk@lexon.ws> To: <timebandit001@gmail.com>; "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com> Sent: Monday, February 21, 2005 12:29

I am having some problems getting TelIax service to work with *. Outbound calls work just fine. When I try an inbound call the phone rings and there is no audio. Upon further investigation "iax2 show channels" indicates that the codec is "unknown" The provider confirmed that they are set for ulaw and so am I. Does anyone have an idea what could be causing the codecs to

============= SJphone Log ============ Outgoing SIP session Respondent: (sip:8612@192.168.2.2) Remote client: Started: May 26 16:33 Accepted: no Ended: May 26 16:34 End reason: Call rejected: 503 Service Unavailable =============== Asterisk Debug ================ Executing Dial("SIP/2201-a83e", "IAX2/<FWDNUMBER>:@iax2.fwdnet.net/612|60|r") in new stack --

All, I run a bunch of (well 20+ actually) Asterisk boxes at home, work, friends and the lie with our own dialplan in the form 8EEXXXX where 'EE' is the exchange number and 'XXXX' is the extension number. This arrangement has been in for 2+ years and worked well with a central box (asterisk.thorcom.net) acting as the routing hub and SIP exchange point with various public

I'm new to this list, and I apologize if this is an already answered question, but my Google-fu was not strong enough to find the answer if it was. I'm having a problem with DTMF on incoming IAX calls. For the first few seconds of the call (between maybe 1 and 15, it varies from call to call) everything works fine. After that I continue get DTMF_E messages from the remote IAX server

Hi All, I am using Asterisk 1.4.26.2 and I am getting the following problem making connections to this server. My other servers are Version 1.2.x which have no problems and this 1.4.26.2 server can call the other 1.2.x servers. The error is: chan_iax2.c:4251 handle_call_token: Call rejected, CallToken Support required. If unexpected, resolve by placing address 192.168.25.250 in the

Hi, My termination with sixtel stopped working, is it something I did or anybody else is having the same problem. I am attaching log: *CLI> -- Executing GotoIf("SIP/300-fbe0", "1?4") in new stack -- Goto (macro-dialout-default,s,4) -- Executing GotoIf("SIP/300-fbe0", "1?6") in new stack -- Goto (macro-dialout-default,s,6) -- Executing

Hi All, Sorry if this has been covered in the past; I've tried searching the archives, but haven't had any luck finding a similar problem. Basically I have problems when using IAX2 (which I now understand is just IAX). I have three IAX connections setup - VoicePulse, IAXtel, and an Asterisk IAX<->PSTN termination provider here in Sydney (ATP) If I try to use the CVS STABLE version

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) ->

I downloaded iaxComm and get up my iax.conf file and the extensions.conf. Here is the out but from CLI in iax debug. What did I forget to do??? Rx-Frame Retry[No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: REGREQ Timestamp: 00001ms SCall: 10489 DCall: 00000 [192.168.50.66:4569] USERNAME : 100 REFRESH : 300 Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 001 Type:

I know in advance maybe I'm overlooking something stupid, however I'm really lost and cannot find the solution... Situation: - asterisk-1.2.13 on a linux box with no iptables active. - one iax2 peer defined - one wildiax phone running on my laptop the soft phone is configured to connect & register on asterisk, however, I cannot get it working. What am I missing? Please help!!

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

I'm taking inbound calls on an * server, then transferring them to a second * server where they join a MeetMe conference. If I have 'notransfer=yes' set on the first * server it works fine, but if I allow the transfer (call then shifts to be between the DID provider and the second server), the call is dropped 3-5 minutes later. There is no firewall on my end, and the two

I have something going on that I don't fully understand after a weekend of looking for answers. I have an iax account with Tesco that works flawlessly with the Zoiper client - but is giving me trouble with inbound calls in Asterisk 1.6. After some playing I have ended up with an iax.conf file that looks like this: [general] calltokenoptional = 77.75.0.0/255.255.248.0 maxcallnumbers = 16382

I run a phone in a remote office using the IAX2 protocol. It mostly works fine; except that every 5 mins it loses connection with Asterisk, before reconnecting 30 seconds later; rinse & repeat. Using the IAX2 debugging, I'm seeing this a lot: Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: POKE Timestamp: 00018ms SCall: 04050 DCall: 00000

Is there any documentation on IAX RSA authentication because I followed http://www.voip-info.org/wiki/index.php?page=Asterisk+iax+rsa+auth and it's not working... Asterisk 1 : -r--r--r-- 1 root root 272 Aug 25 10:34 server2.pub -r-------- 1 root root 963 Aug 24 19:38 server1.key Asterisk 2 : -r-------- 1 root root 963 Aug 24 19:53 server2.key -r--r--r-- 1 root root 272 Aug 25 09:02