similar to: IAX CALLS ONCE MORE

Howdy, I recently saw something strange with a call between *'s over IAX2. There are actually 3 *'s involved. The setup is like this: SIP phone ------(ulaw over LAN)------ *1 -------- IAX2 (ulaw over Internet) ---------*2--------(GSM over Internet) -----------*3--------(ulaw over LAN)------ SIP phone Now what is shown below is the Asterisk in the middle, that is doing the

I have just started working with Asterisk and have run into a road block concerning IAX and an inbound DID from callwithus.com. I am getting nowhere and I don't really know how to isolate the problem. The asterisk version is 1.2.7 on ubuntu, sits behind a firewall with iptables. I can connect and make a call to other internal extensions using zoiper and iax. When I try and use the number,

I'm new to the Asterisk software but have successfully set it up to make and receive calls using FXO cards, voicemail transfer etc. I can successfully call the Digium test IAX using the examples provided. I have signed up for an IAX tel account and got a number. The extensions have been set up as per the examples from IAX tel. However when I try to place a call this is what I get: --

which ir is the * channel? I have decleared 1 peer one in veracruz: [aullox_gdl] type=peer username=aullox_gdl host=200.67.99.127 and aat gdl I have: exten => 200,Dial(IAX/aullox_gdl@200.64.35.58/200@aullox) 200.64.35.58 is my ip 200 is tehe xtension and aull is the context and at veracruz i have: [aullox] exten => 200,1,Wait,2 exten => 200,2,Playback(transfer,skip) ;

Hi, Why do I get registration refused errors with Asterisk and voip providers? I did everything correctly and every provider I signed up with gives me that error: Dec 24 15:30:13 NOTICE[-1254995024]: File chan_iax.c, Line 3955 (socket_read): Registration of 'in-STn46BoD89' rejected: Registration Refused Dec 24 15:30:13 NOTICE[-1298793552]: File chan_iax2.c, Line 4389

Asterisk 1.2.7 RedHat 9.0 I frequently have the need to redirect calls that come in on a DiD provisioned by my ITSP, back to the ITSP so that they can terminate the call on the PSTN. For example when an external call comes in, I often have to send it to a cell phone. I believe that this is referred to as "hairpinning" the call. I do this by answering the incoming call and then I use

Hi all I've been trying to make * work with IAXtel to no avail, all seems ok in the config but am not getting anywhere This is what I'm getting from console (user/pass/dest # changed for obvious reasons): DEBUG[1133735216]: File chan_sip.c, Line 3841 (check_user): Setting NAT on RTP to 0 DEBUG[1133735216]: File chan_sip.c, Line 4891 (handle_request): Check for res for phone1

Hello I am still having problems with IAXTELL and FWD configuration. I get the following when I dial 17009965342 which is set as an example to dial to FWD people. 1+700+99+ 5 digit number. I have placed XXXXX where my passwords are. CLI> Executing Dial("Zap/14-1", "IAX/abatista:xxxxxx@iaxtel.com/917009965342@iaxtel") in new stack -- Calling using options

Hello, Hope everyone is enjoying their holiday! We setup two asterisk servers (From CVS on Wednesday) and set up IAX between the two. Right now they both reside on a switch with a static 192.168.0.x IP address. The first Server is .5 and the second is .30. Our dialplan seems to be working, however on the console we get a flurry of NOTICE and WARNING messages. NOTICE[1116941120]: File

Hey all, I was playing around with IAXTEL last nite and have outgoing calls working a treat. I'm sure I woke a few people up in the US with my annoying test calls. :) Anywayz, incoming calls are a different matter. I have a NAT firewall my * box is sitting behind and the server 'appears' to have registered correctly with IAXTEL. Thing is, when I try and call my 1700 number

Asterisk 1.2.7 RedHat 9.0 I frequently have the need to redirect calls that come in on a DiD provisioned by my ITSP, back to the ITSP so that they can terminate the call on the PSTN. For example when an external call comes in, I often have to send it to a cell phone. I believe that this is referred to as "hairpinning" the call. I do this by answering the incoming call and then I use

Ok so it seg faults when I try to dial out through IAX(voiptalk.org), ofcourse it doesn't if I remove allow=speex :P ---- (gdb) run -c Starting program: /usr/sbin/asterisk -c [Thread debugging using libthread_db enabled] [New Thread 16384 (LWP 28283)] [New Thread 32769 (LWP 28285)] [New Thread 16386 (LWP 28286)] [Thread 16386 (LWP 28286) exited] [New Thread 32771 (LWP 28287)] Asterisk

I constantly get the following error messages in /var/log/asterisk/messages: Dec 8 10:52:57 WARNING[1009521664]: File chan_iax.c, Line 3324 (iax_ack_registry): Received unsolicited registry ack from '192.168.0.1' Dec 8 10:52:57 WARNING[1009521664]: File chan_iax.c, Line 4181 (socket_read): Registration failure Where 192.168.0.1 is another asterisk server. Below are the local and

Hi, I would like to express my appreciation to your big efforts. I am enjoying Asterisk very much. My Asterisk works very well, but I encountered a segfault at voicemail after pressing # to end the recording. Please see the log below. My asterisk is running on RedHat 8. When booting the Asterisk, I found a WARNING around IAX, and it says "Unable to open IAX timing interface: No such

Hi, I created an iaxtel account, and was given a password containing an "@" character. The directory pages imply that they change the web login password only. How do I reset my IAX password so that it is usable in the iax.conf file? Thanks, Steve

Hi Folks, Just wondering if anyone can give me some pointers, I'm configuring Asterisk to talk to FWD's new IAX service. The asterisk server is behind an iptables NAT Firewall, with port 5036 forwarded: $IPTABLES -t nat -A PREROUTING -p udp -d $EXTERNAL_IP --dport 5036 -j DNAT --to-destination 172.16.20.200:5036 I can make outgoing calls just fine, but when I receive an inbound call

Hello, I am new to Asterisk, so RTFM answers welcome too (just include the FM's link :). I'd like to build a simple dialout server based on Asterisk. I installed 0.4.0 from package (a Debian SID machine, "server"). The client is gnophone (a Debian SID machine too, "client"). My modem is a GVC 56k voice modem connected to the server's serial port. I modified

Hi, Recently I am encountering an authentication error when making a phone call between Asterisks. That call is intended as follows. (1) SIP_phone2 to Asterisk#2 (2) Asterisk#2 to Asterisk#3 (3) Asterisk#3 to SIP_Phone3 At (2), that is transferring a call such as -- Executing Dial("SIP/211-6da6", "iax/k0.dyndns.org/302") in new stack -- Calling using options

I have been playing around with asterisk for a week or so now and haven't had too much trouble getting things to work but one thing seems to puzzle me. I have been patient hoping that there was a configuration error on the server or that the toll-free gateway was down but nothing has changed. I have the following configuration context for IAXTEL: [iaxtel] exten =>

Help, I'm stuck. Lost in the woods. I have one Asterisk running on FreeBSD outside on the Wild Internet. One on the safe inside, behind a NAT firewall. The inside server registers with IAX to the outer one and can place calls. The outside one can't register to the one on the inside, since it can't be reached on the private network. Now to my problem: * How do I dial from outside to