similar to: No translator path: iax2 calls not possible

I have two * boxes running two differnet versions of *. Box A is running: Asterisk CVS-HEAD-07/14/04-16:28:29 built by root@asterick.dell.cpu.com on a i686 running Linux Box B is running: Asterisk 1.0.3 built by root@dell.cpu.net on a i386 running FreeBSD I can make a IAX call from B to A but not from A to B. When I try to make a call from A to B I get these messages: Feb 21 12:48:12

Hi everyone, I just got myself a IAXy device and am trying to integrate it to our asterisk server. I configured the IAXy and it is registering and I get a dial-tone. If I try calling another SIP device, and I get "can't translate IAX2 to SIP" How can I make my IAX device communicate with a SIP device (and vice-versa)? Here's what the log says: -- Executing

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

I have a problem with my asterisk and IAX2. It seems that I do not have it or it's broken. I have been trying to connect to another asterisk server and I was thinking I am setting it up wrong. But I am getting this on the asterisk CLI for dialing out via IAX2 even to IAXTEL which is now not working. I used to be able to use it! But that was with IAX not IAX2. Here is the error I get.

Asterisk 1.2.7 Redhat 9 I have DiDs from two different ITSP both set up as IAX2. Each one works when it's the only one in my iax.conf, but when I have them both defined in iax.conf at the same time, only one will work. My iax.conf is provided below. Any ideas how to fix? I'd like to use both DiDs! Thanks, H My iax.conf is below. When I dial the DiD provided by ITSP_B, the other

Hi, I am using the Grandstream HandyTone 486 as a SIP Adapter with a regular phone. Asterisk configuration is listed below. When I attempt to place a H.323 call, I receive the following errors: - Executing Dial("SIP/2000-3029", "OH323/##########@xxx.xxx.xxx.xx:1720|20") in new stack Aug 13 09:13:03 WARNING[20497]: channel.c:1806 ast_request: No translator path exists

Hi, all. I have problems with iax2, when try to communicate with one third server, asterisk reports the following errors in server's, could help me? Server A it speaks It with C in iax and Server B it speaks with D in iax, but Server A it does not obtain to speak with B in iax, reports the following error in server B "chan_iax2.c:5749 socket_read: Host 200.xxx.xxx.xxx failed you

I have successfully configured two servers and I am now trying to connect via iax. When I attempt to call from one ext, 2006(server viop1) to extension 3006 (server voip2) I receive a timeout or "call failed 403 forbidden. The information I am receiving from the console is below. Apr 26 10:53:32 WARNING[311313]: channel.c:1745 ast_request: No channel type registered for 'IAX'

Hello I have created an iax exten in my iax.conf file: [300] type=friend username=300 secret=*** context=default host=dynamic callerid="some name" <300> auth=md5 Then in my extensions.conf I have: exten => 300,1,Dial(IAX/${EXTEN},20) exten => 300,2,Hangup I can dial from iaxComm (a soft IAX client) and that works fine. But when I try to dial 300 get: WARNING[22077]:

Hi all, I've got something strange, that got me searching for quite awhile. Configuration as followed: Linphone on a laptop, that is connected via openvpn to a proxy. That proxy is connected with iax to another asterisk. On the second one i have several hard and softphones. Behaviour at first glance: >From the softphone i can allways set up a connection, But the otherway round fails 9

Hi, I'm trying to set up a basic FXO <> SIP gateway. That is, I want calls from my SIP phone to simply be dumped onto the POTS line. My (entire) extensions.conf is: [from-sip] exten => _9NXXXXXX,1,Dial(ZAP/1/${EXTEN}) and my zaptel.conf is: fxsks=1 loadzone=us defaultzone=us and my zapata.conf is: context=incoming signalling=fxs_ks echocancel=yes

Hello list When i'm trying to dial into our pstn the following errors occure: -- Executing Dial("SIP/snomsip-dbd0", "/2100") in new stack Sep 9 10:02:22 WARNING[59409]: channel.c:1901 ast_request: No channel type registered for '' Sep 9 10:02:22 NOTICE[59409]: app_dial.c:715 dial_exec: Unable to create channel of type '' == Everyone is busy/congested

Hi, Could someone help me on configuring a H.323 trunk. I am trying to set up the following scenario: [SIPphone(2004)]--[asterisk/oh323/asterisk-oh323]--H323Trunk--[PBX]--[H323phone/(8004)] I am using the following versions: Linux CentOS 3.3/2.4.21-.EL.co asterisk 1.0.1 pwlib_1.5.2 openh323_1.12.2 asterisk-oh323-0.6.3b Calling from Asterisk (2004) to the

What I am trying to do is the following: A call is sent to the * box via a SIP invite. The * box answers via an IVR menu system with " Enter the extension you want to dial" so I enter in my 5 digit extension and get the below message. Jan 18 10:10:03 WARNING[-1380238416]: channel.c:1860 ast_request: No channel type registered for 'SIP)' Jan 18 10:10:03 NOTICE[-1380238416]:

Hello list, I've got a problem (maybe only a problem of understanding how * works) with IAX and WaitExten. To simplify the problem I've brought it down to the following scenario: - 3 Asterisk Server A,B and C (central). - A and B both register with C. Now I want to be able to dial an extension at A to become connected to C and there I want to dial an extension to become connected to B.

I think I have it configured properly. ztcfg -vv shows it as channel 1 and zttool shows it as OK. But I can't dial out. When I try, it shows it arrive in teh right stack, but then issues the following errors: channel.c:1676 ast_request: No channel type registered for '{PSTN-1}' app_dial.c:554 dial_exec: Unable to create channel of type '{PSTN-1}' = = Everyone is busy at

I have been running af Asterisk server Version 0.7.2 for a while now But I I wanted to upgrade my version to the new 0.9.0 or the CVS 1.0 Stable. But when I install one of the new asterisk servers I having lots of troubles with the IAX connection between my servers. When I start the 0.7.2 asterisk server it shows me something lige this == Parsing '/etc/asterisk/iax.conf': Found ==

I've got IAX2 setup between two servers with this config: I have two servers on a switch: asteriskm is 192.168.0.160 and asterisk1 is 192.168.0.161 asteriskm has a Sangoma T1 card in it. I want to route calls from asteriskm to asterisk1 which will run an AGI IVR for the call. Config is below, but my problem is that 90-95% of the time when I start asterisk on the two servers I get the

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!!

Please can someone help me with the following two error messages: Error 1. I have loaded the Zaptel dirvers and everything is ok with ztcfg. I have configured Zapata.conf and everthing looks good but it apears the Zap channels dont load when starting Asterisk. When I make a call to one of the fxs port I get the following error message. -- Executing Dial("SIP/39-b204",