Apparently iax is better is it possible to use iax with voipcheap.com ?
IAX is better for what..? IAX stands for Inter-Asterisk-eXchange and is used for talking between/to/from Asterisk boxes. However, while it is theoretically easier to set up, in that it only uses a single port (usually 4569) rather than the myriad of port numbers used by SIP, it isn't necessarily "better" as such. I and a friend in North Wales are currently scratching our fingers down to bone level trying to figure out why an IAX ATA at my end is suffering no end of problems connecting to his Asterisk PBX whereas a similar SIP unit is working fine..! Ivor
"David Quinton" Yes. We're beginning to think it's a firmware bug in the IAX ATA - http://www.atcom.cn/En_products_AG188.html as someone else is having exactly the same problem connected to a totally different Asterisk box. The device is very new on the market, we're waiting on a reply from the manufacturers. Ivor
IAX is very proprietary. Even though the source code is open, it is still a specific to asterisk thing. Tim
Maybe that's why it's called Inter-ASTERISK-eXchange ;-) If anyone's interested, the system we're using is CNET (see www.ckts.info) Ivor
Have you tried the manufacturers forums? http://bbs.pa1688.com/ (I know it's not a PA1688-based ATA, but that's just the url of the site).
Yes, already looked there. Nothing specific to our problem. It doesn't help that CNET isn't exactly a mainstream use of VoIP..! Ivor
Try IDEFISK from http://www.asteriskguru.com. If this works then unfortunately it is your ATA. Best regards.
Thanks, I'll give that a look. The problem is, some of the trouble we're having is intermittent, i.e. a call attempt will fail two or three times, then succeed on the next try. Ivor
Is there a qualify= line in the iax.conf entry on the remote host? Are you registering to the remote host or are you using static hosts?
I don't know about qualify= as I'm not at the remote end and my knowledge of Asterisk/Linux is virtually non-existent..! However I will ask and let you know. What should be in that line..? Using registration as you would to a normal VoIP provider. Incidentally, I've tried the DIAX softphone on my laptop and that works fine. Ivor
Asterisk in most configurations logs most or all of what it does in /var/log/asterisk. You can also increase debugging with 'iax2 debug' from the console. Annoyingly enough it's not possible to debug an individual IAX peer or trunk in the way you can with SIP :-(
Asterisk in most configurations logs most or all of what it does in /var/log/asterisk. You can also increase debugging with 'iax2 debug' from the console. Annoyingly enough it's not possible to debug an individual IAX peer or trunk in the way you can with SIP :-(
Asterisk in most configurations logs most or all of what it does in /var/log/asterisk. You can also increase debugging with 'iax2 debug' from the console. Annoyingly enough it's not possible to debug an individual IAX peer or trunk in the way you can with SIP :-(
In this situation, normal is only defined by your setup, if you both have static IPs then you wouldn't want to register. (registrations are quite uncommon with iax, so it wouldn't surprise me if the registration code in the ATA hasn't been exhaustively tested). The only VoIP providers I use that will use iax to terminate to you, (gradwell and voiptalk) do not support registrations. (afaik). It would also be prudent (if the ATA supports it) to use RSA authentication. (if you don't know, ask).
