[WXC/VFX] Trixbox: Callers told user busy, goes direct to voicemail

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nickwallingford

24 posts

Geek

#32809 24-Apr-2009 07:19

I've had to do a re-install of trixbox after (ahem!) breaking everything...

I've followed Tony's guide for the most part, with only minor deviations (pan.wxnz.net and a few variations based on work by josh, and the initial posting that comes out when signing up from WxC).

I've got a trunk, incoming and outgoing set up, presumably as I had done before - pretty vanilla setup. Outgoing calls work fine, but all callers into the system get the message that "The person at extension 100 is on the phone. Please leave your message..." etc, and the invitation to leave voicemail. I've found other users with similar problems but more often related to phone type. Mine is going only to analog phones through a PAP2T. Could it be that this is not 'registering completely' as one suggestion had it? It appears as being registered OK in the status. The fact that outgoing calls are OK seems to indicate that the registration of the trunk is working properly, doesn't it?

So incoming calls do not ring the phone, and the users go immediately to voicemail.

I'd appreciate any help, and happy to post any settings. As a start, what I've got in the PEER section of the trixbox trunk is:

port=5060
dtmfmode=rfc2833
disallow=all
allow=ulaw&alaw
type=peer
fromdomain=pan.wxnz.net
fromuser=(myphonenumberwithoutleadingzero)
host=pan.wxnz.net
insecure=invite,port
canreinvite=no
nat=yes
secret=(*****assupplied)
username=(******assupplied)

USER CONTEXT is AuthID (that is literally the word 'AuthID', eh? Not the AuthID which is the username above???

Nothing in USER DETAILS

Register string is:
myphonenumberwithoutleadingzero:(****assupplied):(****assupplied)@pan.wxnz.net/myphonenumberwithoutleadingzero

-----------

Calls to my number are obviously getting to my trixbox (prompts are 'mine', and I can see the call pickup on the initial PBX settings screen).

Any help would be very much appreciated. I'm at work most of the day (until beer o'clock) so can't do config changes until later, but I'm willing to try them all to get it working...

Nick

mjb

996 posts

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Trusted

#208941 24-Apr-2009 08:56

The PAP2T is registering with Trixbox properly? Do you have a SIP peer set up for it in sip.conf? Does the dialplan correctly direct calls to the extension to the sip peer?

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nickwallingford

24 posts

Geek

#208950 24-Apr-2009 09:31

I'm not at the PC just now, but from my understanding, the PAP2T does seem to register. The lights are 'as they should be', and on the status screen there is a peer showing as extension 100. Again, with it not in front of me, on the opening PBX settings screen, where at the bottom/middle you can see green bars to show registrations, it appears to be registered OK.

And the fact that I can ring out OK - would I be able to do that if the PAP2T was not registering correctly or now?

As a relative newcomer to trixbox and asterisk, how best can I confirm "Does the dialplan correctly direct calls to the extension to the sip peer?" Does the fact that I'm getting the "The person at extension 100 is on the phone" message confirm that, or would that be generated anyway?

I guess as much as anything, what I'm looking to learn is "What debug/diagnostic can I readily do?" I did use the asterisk -t at one point to confirm a few more things, but can't remember the best way to confirm such as your questions to me...

Nick

sbiddle

30853 posts

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Biddle Corp

Lifetime subscriber

#208951 24-Apr-2009 09:43

If it's Asterisk giving you the message then the problem lies with the registration of your PAP2T with Asterisk.

nickwallingford

24 posts

Geek

#208953 24-Apr-2009 09:48

So is there a concept of partial or inadequate registration of the device into asterisk, or is it al all or nothing sort of deal?

I ask because I have worked at least partly from the basis that as I can ring out through the device OK (that is, the same phones plugged into the PAP2T) that gave me some confidence that the PAP2T had registered into asterisk OK.

Nick

mjb

996 posts

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#208957 24-Apr-2009 09:54

nickwallingford: And the fact that I can ring out OK - would I be able to do that if the PAP2T was not registering correctly or now?

No, the registration is simply so that Trixbox knows where to send calls for that sip peer - the PAP2T knows where to send outgoing calls, as it has your Trixbox server configured as it's peer.

It's the same with the registration to your ITSP - that's just so they know where to send incoming calls for your number. For outgoing calls, Trixbox knows where to send them because you've configured where in the sip peer.

nickwallingford: As a relative newcomer to trixbox and asterisk, how best can I confirm "Does the dialplan correctly direct calls to the extension to the sip peer?" Does the fact that I'm getting the "The person at extension 100 is on the phone" message confirm that, or would that be generated anyway?

I guess as much as anything, what I'm looking to learn is "What debug/diagnostic can I readily do?" I did use the asterisk -t at one point to confirm a few more things, but can't remember the best way to confirm such as your questions to me...

I must apologise, I'm not a trixbox user - I just installed asterisk from my linux distro on my server at home.

The "is on the phone" message is just because the VoiceMail application is being called with the 'b' (busy) argument, rather than the 'u' (unavailable) argument. The dial plan (extensions.conf) will probably be more enlightening about why that is - when 100 is called, usually it'll use "Dial" to send the call to the SIP peer, but if the mapping to that SIP peer isn't correct, the call will not work.

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mjb

996 posts

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#208962 24-Apr-2009 10:01

To give you an example, my incoming context is like this:

[globals]
CORDLESS=SIP/ATALine1
OFFICE=SIP/Phone1
LOUNGE=SIP/Phone2

HOUSE=${CORDLESS}&${OFFICE}&${LOUNGE}
VM_HOUSE=20

....

[incoming-from-iTalk]
exten => s,1,Dial(${HOUSE},20,tkr)
exten => s,2,Goto(s-${DIALSTATUS},1)
exten => s-NOANSWER,1,Voicemail(${VM_HOUSE},u)
exten => s-BUSY,1,Voicemail(${VM_HOUSE},b)
exten => _s-.,1,Goto(s-NOANSWER,1)

In sip.conf, I have peers "ATALine1", "Phone1" and "Phone2" that equate to the SIP devices in the house.

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nickwallingford

24 posts

Geek

#209141 25-Apr-2009 09:02

PROBLEM: New install of trixbox would not ring phone, but would send user straight to voicemail, telling them the user on that extension was on the phone...

RESOLUTION: PAP2T was not registered properly in some way into the asterisk. Reset it and re-entered my details and all is OK now...

mjb and sbiddle - thank you both for helping me. Your suggestions led me to look more closely at the registration of the PAP2T's registration. While it appeared to be OK from what was showing in the trixbox status, etc, there will have been something wrong here.

I went to the asterisk CLI (asterisk -r) and turned on debugging (sip set debug) then monitored the log (tail -f /var/log/asterisk/full) to watch what would happen.

The PAP2T was sending the "always busy" signal to asterisk when it negotiated...

I plugged a phone into line 1 and did a reset (****73738#), got a new IP (****110#), went to that IP and put in my misc details, giving it a fixed IP, telling it to proxy the IP of my asterisk server, etc.

And all was resolved!

Again, thank you for offering such good suggestions - they were spot on to lead me into what I had only sort of suspected based on various searching before I asked.

With each time I break this stuff, I learn. Or at least I *hope* so. It sure isn't going to stop me from trying!

Nick