Telasip debugging...

[ListerD]

Having an issue with Telasip and need some help tracking it down.

Gene and I have been emailing back and forth on this and we're both kind of stumped as to where to begin looking at this.

On his end it looks like I'm not logged in, yet I can dial out on the telasip trunk with no problems at all.

Incoming calls to my DID number get the message that "extension xxxxxxx (my DiD) is not available". This is coming from the telasip side, and I have to agree it does look like I'm either not logged in or in DND mode.


Looking at my IAX peers with Asterisk info I see:

telasip/myaccount name 4.79.19.58 (S) 255.255.255.255 4569 OK (56 ms)

I see no errors listed in the CLI.

Next step?

[ListerD]

Interesting... From the CLI "iax2 show registry" shows no connection to telasip.

Is there a manual string to make the connection from the CLI? I'm not finding it under the help or online.

[AlexanderBell]

What does your Trunk info for TelaSIP and Registration string look like?

[ListerD]

Gene just emailed me that it looks like an MD5 authentication failure. So I'm waiting to see what he wants to do next.


Outgoing settings:

host=gw3.telasip.com
qualify=yes
secret=pass*
type=peer
username=username


Incoming Settings:

User Context: username

User Details:

context=from-pstn
host=gw3.telasip.com
type=user

Registration:

Register String: usernameass@gw3.telasip.com


*The password has a dot (.) in it, which is interesting. I was raised in *nix as that is a no-no in passwords...

I'm wondering if we shouldn't specify auth=rsa but I'm not sure in which context it should go into.

[ListerD]

(insert Homer_Simpson_32-Doh_audio_file)

It was my password... I had it correct in one spot, but dropped a character when I added it to the registration string! ops: :evil:

And the really stupid part is I've read and reread those files in vi and in AMP and never spotted the typo. I don't feel too bad because Gene didn't catch it either until about 15 minutes ago . :wink:


OK - so I have nothing but glowing recommendations for Telasip! Gene busted his rump helping me on this.

[AlexanderBell]

There is no better provider out there. Trust me. I've tried (more like endured) dozens.

[tdar]

OK I have a very strange question! I set my * server up last night and I have it working for both incomming and outgoing calls...EXCEPT when my PTSN landline calls the telasip line! When you call from the PSTN line to the telasip DID...The call is receved by my server and routed to the correct ring extention. But that extention reports that it's busy and the call goes to WM. No other caller get this action! To make it even stranger it worked this AM 1 time and now it always goes to WM...I'm lost HELP!!!

[AlexanderBell]

I wish I had enough fingers and toes to count the number of people that get screwed by AMP's [from-sip-external] every day! One of the worst default settings anyone could have possibly thought up. The solution for the benefit of future Google searchers is to comment out all three lines above and add the following:

exten => _.,1,Goto(from-pstn-timecheck,s,1)

Then your incoming SIP calls will actually be processed and ring just like incoming PSTN calls instead of callers getting a fast busy or voice mail welcome on every incoming call.

[ListerD]

Just to clarify - he's speaking of the entry in extensions.conf.


I have to say, you just saved me a HUGE headache! Now not only does the DID route now work properly, my ring group also works! My Grandstream BT101 would always ring busy as tdar pointed out and this broke a DID route to it as well as a ring group.

As the younger kids would say - You da' man!

[tdar]

My problem is a bit different....It's only one Phone number that will not ring ....all other numbers work fine!