similar to: Multiple IAX client behind a NAT

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

Folks, It seems like the * v 0.9 and iaxcomm won't speak to each other. Is there another IAX2 client that is usable under Linux (Debian preferred)? Thanks, Tim --

Hi List; I heared that IAX is good for NATing issues, but I do not know if it can help me in that senario: I have two Asterisks machines in different sites and both are behind NAT (both have private IP address), I need to link these two asterisks with IAX trunk (if it help really in such senario), but I do not know if it will work without doing special routing settings on the router (like

I'm trying to set up * server behind NAT. The box is set up as DMZ in my DSL router, i.e. all incoming connections without explicit port mapping are forwarded to *. So far I'm unable to get this setup to work for either IAX or SIP (tried IAXComm & XLite softphones on public IP address). Data seems to come in fine (IAX/SIP debug shows message interaction taking place), but there is no

Hello! I'm searching for a way to call ATA (IAX or SIP) that is not registered with any server or proxy. Is it possible to make such a call from a softphone to an ATA just with IP? Something like (sip:// or iax://)1111@210.12.34.45 (where 210.12.34.45 is ATA's public ip)? Regards, CuPoTKa.

Hi all, I've had a strange problem this morning and I know someone who has reported exactly this problem to me too last week: - I've setup a new server running Asterisk 1.2.4. Currently there is no Zaptel hardware install (but there will be soon). This server is behind a NAT router on an DSL line. The remote IAX server on the Internet (which handles the call termination / origin)

i am looking at iax to see if it is applicable to my needs. i would appreciate any corrections of what i think i have understood but probably have not. iax uses udp and traverses nats. neither of these seems useful to me. i loathe nats, and udp is not well-behaved in the sense of congestion avoidance. trunking will save some bytes in flight iff one has four or more streams moving between two

Hi folks, Ok, I've seen this question go unanswered on the mailing list, and I assume it's because no one had the heart to break the bad news to the guy asking, but be honest with me, I can take it. At this time it's flat impossible to have multiple IAX phones behind a NAT without using an * gateway because there's no way to have a client listen on a port besides 4569. Is

Hi, after reading about it on the list I decided to set up a Free World Dialup account. For those of you who don't know, that is a sip proxy where you and your friends can singn up free and then you can just connect to it with any sip client and call anybody that is registered for free. Pretty much like iaxtel (I belive that was the name of it) for the iax protocol. It even supports clients

Current dev cvs install on two systems. System A is behind a SonicWall firewall, and system B is on a registered IP address. (System B has multiple iax links that are fully functional to multiple locations.) System A is correctly registering with System B, with no special firewall rules. Should System B be able to take advantage of the "registration" to send iax/gsm calls to System A

Hi all, I've searched the wiki, and my basic assumption at this point is to run multiple IAX clients behind NAT I need to specifically code each client to use a different port and then setup that port to be forwarded from the NAT router to their private IP address. At the moment, I can't seem to get more than one IAX client registered behind NAT... am I correct in my above

Dear List I have notice this kind of problem between my two * box. My scenario is : Iax GSM IaxClient----->PBX1------------>PBX2-->TDM today CVS Stable V1 I use as Client FireFly with IAX2/GSM and try to call my PBX1 this server call PBX2 to terminate the call trought a TDM line (TE410P) but after PBX2 join the two call i can see the log below from my PBX1, i can speak for

Hello all, I am having trouble getting my IAX based Voip provider setup. Any pointers are welcome. So here is the deal. I am registered up and I can make outgoing calls but incoming calls fail. Configs all look good I thought. My PBX is behind our firewall with a direct NAT of one to one for an external IP. IAX port is forwarded UDP and TCP to the internal IP. * shows good registration and

Hi, I am problems with iax2, when try to communicate with one third server, asterisk reports the following errors in server's, could help me? Server it says It with C in iax and Server B it speaks with D in iax, but Server 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 authenticate sipspo

Hi All, I know, IAX is NAT friendly, but... I have a problem running gnophone from a box behind NAT firewall. I can register gnophone with * through NAT, but when I try to make a call it instantly disconnects. CLI iax show peers command tells me that peer is unreachable. However this peer is registred. Gnophone also tells me that it is registred. It seems that registration handshake has

I am trying to make 1 soft SIP UA behind NAT connect to a public hard CISCO UA via a public asterisk server. The CISCO UA can hear the voice from the SIP UA but not vice versa. I do set nat to yes for the soft phone. Any help would be greatly appreciated. Below is my sip.conf [general] port = 8060 ; Port to bind to (SIP is 5060) bindaddr = 0.0.0.0 ; Address to bind to (all

Hi! I have this configuration: SIP client A <-> NAT box A (real external IP) <-> Asterisk server (real IP) <-> (real external IP) NAT box B <-> SIP client B The echo test form any of the clients to the asterisk server is working just fine, even without canreinvite=no. When I try to call from SIP client A to B, wihtout the canreinvite=no in the sip.conf, the call

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

Hi All, I'm using DISA on my * server to avoid overseas toll charges when making calls to Western Europe from my cell phone. I have DISA working with a DID from a VoicePulse Connect account. The outgoing call to Europe is also made via Voicepulse Connect. I see that the IAX media path is bridging the inbound call to the outbound call so that the media stream entirely bypasses my server once

Hi, I am pretty new to all of this but was able to set up an asterisk server and have been able to succesfully connect to asterisk with x-lite as sip client. I have also connected asterisk to FWD (using iax2) and to voipjet (also using iax2). Now I am trying to connect asterisk to Stanaphone. It has to register as a SIP client but I am not being succesful at all. My asterisk server sits behind a