similar to: Help!!!: test Asterisk error: error on IAX1.conf and warning on chan_iax2.c

Hi, I have a wildcard x100p. I just installed asterisk by following step: # cd ../zaptel # make clean ; make install # cd ../libpri # make clean ; make install # cd ../asterisk # make clean ; make install # make samples When I test Asterisk typing # asterisk –vvvvc I find one error and one warning: [chan_iax.so] => (Inter Asterisk eXchange) == Manager registered action IAX1peers ==

Hi I?m a new user and I do test with my hardware . I have a x100p and telephone vozip. And when I run this command asterisk ?vvvvc for to test it . My computer show it ?warning? [chan_iax.so] => (Inter Asterisk eXchange) == Manager registered action IAX1peers == Parsing '/etc/asterisk/iax1.conf': Not found (No such file or directory) Apr 2 07:45:12 ERROR[16384]:

G'day list, I am getting a lot[1] of traffic on my Internet link, ICMP messages from 69.73.19.178 telling me UDP port 5036 is unreachable (this IP address belongs to iaxtel.org). I see from the wiki that IAXtel supports only IAX2 from December 2003. Fine, however it looks like my * still wants to try and register using IAX1, and I can't find how to turn this off. This situation is

I was just noticing that my asterisk box is trying to register with my providers on both IAX1 and IAX2. They do register on IAX2, but it seems like it keeps trying to register with IAX1 also. Is there a way to disable this behavior, or am I just misunderstanding it. I just want to make sure I'm not wasting processor time and bandwidth trying to register IAX1 when it is already

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 ==

hello, I have been using asterisk now for about 2 years now on a RH8.0 it is our main call gateway. I have on the box 3 T1 TDM cards connected to 2 Rhino channel banks (FXS) and 1 CAC Access bank I (FXO) with so many softphones and ATA 186s. It has been working good till today some few hours ago. i just discovered that there were no dialtone on the phones. Asterisk did not spit out any error, it

I'm still using asterisk v. 0.9 I think it is using IAX1 as the ixa.conf is set port=5036 Can I register with a provider who is using IAX2 ? When I set it up and run: iax2 show registry - it is not displaying any registered provider. -- #Joseph

Just in case anyone else is looking for this information, I'm posting the answer here. I do need to double check if this works with asterisk >= 0.9.0. I've heard rumors that IAX1 support has been removed in newer versions. ---------- Forwarded Message ---------- Subject: Re: [Asterisk-Users] Disable IAX1 Registrations Date: Thursday 17 June 2004 04:43 pm From: "Chris A.

Hi folks Is it possible to turn off the native bridging on Asterisk? I've been hacking about app_disa.c to support account & pin numbers, that tag the calls depending on who logs in..... It all works fine, then dials the destination number they requested. My setup is as follows [ENDPOINT] <IAX1> [MYASTERISKBOX] < IAX1 > [TELCOBOX]<>(PSTN) If i dial

Hi, I occasionally experience a one way audio problem when using the IAX1 protocol with a service provider. If I call a DID number in by group provided by the supplier which is redirected to my box, I only get the audio path back from my box to the phone I used to dial the DID number. If I use IAX debug, or tcpdump I only see packets going from my box to the provider when the one way audio

Hello, I have searched the lists, and have seen this message posted numerous times, however there never seem to be any replies to it. I was curious if anyone had figured out the WARNING: File chan_iax2.c, Line 5466 (set_config): Ignoring port for now Everything seems to work just fine, however when I try to change iax.conf port to 4569 it doesn't seem to like that either. Thanks, Brent

<P>Has anyone managed to get Asterisk to work with Glophone/Voiceglo since this posting.</P> <P><A href="http://lists.digium.com/pipermail/asterisk-users/2004-February/036559.html">http://lists.digium.com/pipermail/asterisk-users/2004-February/036559.html</A></P> <P>I've tried copying the config in this listing with no success. </P>

I'm attempting to get two Asterisk 0.4.0 PBXes to communicate with one another using IAX/IAX2 trunks. I've managed to get a semi-functional NAT Firewall working as a PBX (with Asterisk running directly on the firewall itself), but there are issues with bind()ing to various interfaces which is causing outbound SIP issues. To get around these issues, the idea is to do something like

Hi all, I'm using asterisk in a production environment now and this afternoon I got reports complaining that it was not working. Looking at the asterisk console output, I saw it contains lots of error messages as printed below. Unfortunately it is not obvious from the logs as to what started all this. Just before the error messages start, everything seems to be working fine with no problems.

Hi, I am very closed to implement the IAX2 version in DIAX, but still some issues which I don't know how to handle, maybe someone from this list can help me. Trying to register with the * server as in version 1, I get the following in the * console: NOTICE[1150495040]: File chan_iax2.c, Line 2919 (register_verify): Inappropriate authentication received and in the client: Registration

i.e. iax1-------+ iax2-------| iax3-------|-- * -- MeetMe -- shoutcast radio ....-------| iaxn-------+ would be a great way for corporation to broadcast meetings. board meetings live management meetings live sales demonstrations and all interactive. you get the picture. right now i'm thinking of just patching a cable from the * operator console to a shoutcast server.

Hi all I been looking into the whole code strugture of chan_iax and i see there is a option to specify the refresh rate of registrations: But there is no code to actually load this from the config file thus i changed the setting in chan_so.h, and recompiled. But still my refresh rate is 60 sec. I need to get this down to 15 sec (nat /pat firewall issue) any ideas? thanks Liaan

Hello-- Why do I target chan_sip for so much effort? Because, it seems to me, chan_sip is probably the most used channel driver in the asterisk community!! (and, of course, the zap/dahdi driver, is also pretty popular) I haven't had time to follow up on chan_sip, and I probably won't for several months. But, if I had time, here is what I'd do: There are two ways to speed up

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

Today has been the day for satellite questions, apparently, so I'll proxy one out to the rest of the community... I asked this tangentially a month or two ago, but I'll put it in a more blunt way: If you have IAX2 trunking mode experience over satellite, please let us know your experiences with that protocol/transport combination. I've got several people asking about IAX2 and