Sixtel and IAX.cc

[CyClore]

Anyone else having a problem with them? I have two 800 numbers through them and one of them that I don't use much somehow got changed to SIP. They told me they put it back to IAX, but they didn't, I can see that from the CLI. Ever since this morning or sometime last night my main 800 number that I gave my family quit working. I took the fail over off so I could test it and it doesn't even make it to the Asterisk machine. I have verified that I am registered, so I know that is not the issue. This has been running IAX right along and I haven't had any problems with it. Now it won't even connect here.

I need an 800 number for my family as the closest DID for me is still long distance for them and I know it will be a long time, if even that I can get a local did in this middle of the woods town (even though we can get a 10 meg cable connection at home. Go figure).

So... Anyone else having a problem with them? I am thinking of getting another DID from them just to test with.

[CyClore]

I started digging in a little deeper and wondered if someone could explain to me what is going on. This is from the iax2 debug...

: IAX Subclass: ACK
Timestamp: 00747ms SCall: 00004 DCall: 00506 [70.84.157.148:4569]
Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW
Timestamp: 00018ms SCall: 00576 DCall: 00000 [70.84.157.148:4569]
VERSION : 2
CALLED NUMBER : 800XXXXXXX
Unknown IE 045 : Present
CALLING NUMBER : 570662XXXX
Unknown IE 038 : Present
Unknown IE 039 : Present
Unknown IE 040 : Present
CALLING NAME : 570662XXXX
LANGUAGE : en
USERNAME : sixTel
FORMAT : 4
CAPABILITY : 2097151
ADSICPE : 2
DATE TIME : 207114961

Ignoring unknown information element 'Unknown IE' (45) of length 0
Ignoring unknown information element 'Unknown IE' (38) of length 1
Ignoring unknown information element 'Unknown IE' (39) of length 1
Ignoring unknown information element 'Unknown IE' (40) of length 2
Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: REJECT
Timestamp: 00005ms SCall: 00001 DCall: 00576 [70.84.157.148:4569]
CAUSE : No authority found

Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: ACK
Timestamp: 00005ms SCall: 00576 DCall: 00001 [70.84.157.148:4569]


TO me it looks like the inital connection is makeing it to me but I a rejecting it. I guess my question is why? I haven't made ANY changes to my system at all.

Anyone give me a hand?

[isepic]

Originally Posted by CyClore

I started digging in a little deeper and wondered if someone could explain to me what is going on. This is from the iax2 debug...

(truncated)

TO me it looks like the inital connection is makeing it to me but I a rejecting it. I guess my question is why? I haven't made ANY changes to my system at all.

Anyone give me a hand?

I'm not sure what is goign on with you, but I did notice they changed the control panel (on their site) to allow you many different options. I was trying to change from IAX2 to SIP (opened all the ports on my fireall and all that).. decided it wasn't for me, and then changed back. But I've included a screenshot of what I see.. hopefully you'll see the same thing and are able to change it to which ever type of device you want it to be.

It looks like it said "username: sixTel" not sure if that is right, I have my trunk name as sixTel (cap T very important) and my username is my login name that I use to access Sixtels web page. Anyway, I wish you goodluck, and I usually have the best of luck calling them at 10am (PST) or finding Chris on MSN messenger (look on the support page for the username).

[isepic]

One other thing with them, over the past 7-8 days I've had choppy calls with them. I performed some packet testing and found out I needed to change my MTU value down to 1250 in order to get good pings to them (sip.sixtel.net) w/o any packet loss. This isn't the case to my other providers. I have good trace routes to them, everything is well below 100ms all the way - not sure what is going on. I called them and told them I have choppy calls with them, but not with 2 other providers. I since changed my MTU values to ensure no packet loss, and it seems to have helped a bit, but every 3-5-7 seconds I get a cut out. Weird. So I just wanted to say other than this, I've had no problems with them.

[CyClore]

I got their email yesterday about their call quality with iax2. I did the change and now my incoming doesn't work at all. I hate it when they make these changes because anking them a simple question may never get answered because their tech cupport sucks.

To register now they want you to use register.sixtel.net instead of iax2.sixtel.net and they also want you to change to outgoing.sixtel.net instead of iax2.sixtel.net. I am not using their outgoing service, just incoming. I tried all the different possible solutions and I just can't get it to ring in anymore.

Anyone get the A@H machine working using IAX since there little email came out?

[isepic]

I got their email - seems they are having some issues with IAX. Are you using IAX or SIP?

I switched over to to SIP and opened the ports they suggest (in support, under help loo for SIP) and all works fine, BUT, it didn't at first, I cut and pasted, and when I did, the incoming context they use "incoming" or whatever it is, doesn't match mine, so I changed it to from-pstn --- works okay now.

Not sure if that is your prob or not... but I'd start off with the help section for either SIP or IAX that they have in your account settings, also verify the control panel settings (as mentioned above) are pointing to the right one you want (sip/iax2, etc.). [ and if you cut and paste what they have there for you, don't be like me, make sure the incoming context is something REAL on your system ]

[CyClore]

I planned on posting this before now, but I have been under the weather.
I didn't experience any of the choppy audio. I simpley changed from iax2.sixtel.net over to register.sixtel.net and I am working 100%. I like using IAX because my touch tones (DTMF) isok with it so I can check my VM. My NAT firewall plays to much with my SIP connections and they do not decode directly so I wanted to stay with IAX.

All is working 100% now.

[TriRyche]

Originally Posted by CyClore

I like using IAX because my touch tones (DTMF) is ok with it so I can check my VM.
My NAT firewall plays to much with my SIP connections and they do not decode directly so I wanted to stay with IAX.

Now that Gizmo 2.0 has Native Asterisk Support, if properly configured,
would Gizmo's current ability to traverse most NAT Firewalls while using SIP (STUN?)
somehow be able to assist you with your DTMF problem?

Also, refer to these other Voxilla threads -

• VoiceMail won't respond to commands
  DTMF problems...anyone else?
  Outbound calls don't register DTMF Tones AGAIN
  Understanding DTMF

- TriRyche

[rybshik]

Looking for a provider of inbound DIDs which will let me point (forward) incoming calls to any arbitrary SIP URL. Will any of Sixtel.net / Vitelity.net / IAX.CC / exgn.net do it?