[WXC/VFX] Asterisk Feedback

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maverick

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WorldxChange

#17378 24-Nov-2007 09:49

Hi All,

If you have any comments about your experiences with Asterisk on VFX please post them here, would be interested in how you found

A) The service (working / not working)
B) The Features (working / not working)
C) Quality of calls
D) How you found the setup at your end , do you need more info etc
E) Any thing else you may want to add, the good and the bad (were not afraid of a bit of constructive criticism

)

We have already seen some feedback on the support lines and changing existing VFX users over to Asterisk, so we are addressing some things there so your comments are helpful to us in providing the best service we can , so please pass on your comments here.

Phil Moore
Operations Manager
WorldxChange Communications

Yes I am a employee of WxC (My Profile) ... but I do have my own opinions as well

https://www.facebook.com/wxccommunications

mruane

420 posts

Ultimate Geek

#96714 24-Nov-2007 14:30

I am really happy with this new service and call quality is excellent.

One thing that would help is for the sticky thread to be updated with a FreePBX configuration. Using the Trixbox configuration from the sticky, the trunk never appears in FreePBX as an available trunk to use. I have created the trunk in FreePBX and it does work fine. If you want a screen shot let me know.

Cheers Mike

sbiddle

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#96717 24-Nov-2007 14:38

I'd be happy with write up a DIY guide with screenshots but I believe Tony may have already done this.

frio

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#96718 24-Nov-2007 14:51

I've got a crappy guide on the way for Gentoo users too :).

With respect to feedback:
The service is working, and working well. Asterisk is routing calls as required by my crappy little setup, and when the server is down VFX's automated answering machine and such take over which is choice.
As far as I am aware all features are working.
Quality is outstanding. Can't tell the difference between the softphones and a landline, which is perfect. The delay is in the order of a couple ms at best which is fantastic.
Setup was good. As mentioned, a full guide would be nice (Tony's is excellent) but I think the initial difficulty for me was the fact I've never used Asterisk (or a PBX!) before, so learning how that all worked was the hard stuff. I'll be covering some basics in my guide with respect to that. The support from Phil was excellent ^_^.

Overall yeah, very satisfied indeed. I don't really need a VFX line in all honesty, but the opportunity to play around with Asterisk and learn some new skills definitely swayed me. I'm very happy with the results.

maverick

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WorldxChange

#96725 24-Nov-2007 15:48

sbiddle: I'd be happy with write up a DIY guide with screenshots but I believe Tony may have already done this.

Yes Tony has written an excellent addon Guide to setting up a VFX trunk with a nice bit of Tony Hughes humour on the side, it makes it a fun and easy setup, he was just clearing it with us but first hopefully he will be putting up a sticky thread with it shortly.

Yes I am a employee of WxC (My Profile) ... but I do have my own opinions as well

https://www.facebook.com/wxccommunications

tonyhughes

Hawkes Bay
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#96726 24-Nov-2007 15:48

I have already written a full step by step guide for VFX+Trixbox users (a new one, not my older Tribox 2.0 for dummies one)

I am just ironing out some routing bugs before I post it.

It starts with 'insert the CD' and includes 'click OK' , 'click 'Apply configuration changes' and other such basic steps, as well as instructions on performing initial system updates etc.

I have been away on holiday, which has hindered progress!

munchkin

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#96748 24-Nov-2007 19:37

I'd have to say I'm enjoying the improved call quality improvement when compared to the PAP2T (As Maverick said 'It sounds like you're next door') and having the boss ring you at home just to see if everything is working is a nice touch, I have to say.

DribblingBadger

9 posts

Wannabe Geek

#97442 28-Nov-2007 22:00

OK Tony, as you can see from my status I have gone from scratching my head to tearing my hair out about the asterisk dial plan. I have got succesful registration with the VFX gateway, and a couple of SIP phones working that can dial internally, but no incoming or outgoing calls as yet.

I hate to hassle a man I don't even know, but how is that badgers guide to asterisk on VFX coming on ?

cheers

DB

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tonyhughes

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#97454 28-Nov-2007 23:02

From my guide, but please bear in mind I am having some issues with incoming calling. I am not sure where the fault lies (But m positive its not a WxC problem). Please let me know if you get incoming calling working or not in Trix with these settings (and the Trix version you are using). Outbound calling is working great for me, and inbound calls hit my Trixbox, but I cant get them routed to an extension for some dumb unknown reason - and yes, I have a decent valid inbound route).

=============================================

(Where your phone number = 091234567)

• Click on 'Trunks' in the sidebar
• Click on 'Add a SIP trunk'
• Outbound caller ID: 091234567
• Never Override CallerID: Tick this
• Maximum channels: 1
• Trunk name: VFX 1 (Can be anything you want)

PEER details:

allow=ulaw&alaw
canreinvite=no
disallow=all
dtmfmode=rfc2833
fromdomain=as.wxcnz.net ;(dont forget to edit your hosts file as instructed!!)
fromuser=91234567
host=as.wxcnz.net
insecure=invite,port
nat=yes
secret=thepasswordWxCgaveyou
type=peer
username=theAuthIDWxCgaveyou

USER context: theAuthIDWxCgaveyou
USER details: (leave empty)

Register string: yourVFXNumber:PASSWORD:AUTHID@as.wxcnz.net/yourVFXNumber (no leading zeroes on the VFX numbers)

• Click 'Submit Changes'

tonyhughes

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#97455 28-Nov-2007 23:11

munchkin: I'd have to say I'm enjoying the improved call quality improvement when compared to the PAP2T (As Maverick said 'It sounds like you're next door') and having the boss ring you at home just to see if everything is working is a nice touch, I have to say.

Yes, the call quality is brilliant!

joshp

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WorldxChange

#97474 29-Nov-2007 08:26

Hi Guys,

Can you PM me some of the failed call attempts, I'll take a look at what is being sent and what your box is returning to us..

if you can run something such as ngrep (an ethernet monitoring tool..) use the following command:

ngrep -d eth0 port 5060 and host 58.28.20.150 - * note this is a linux cli command.. (eth0 is the network card in question.. use ifconfig -a to determine if this is the correct interface to monitor).. place a call into your asterisk box..

this will give you the sip messages that the ethernet card is seeing.. it should show you what the number is coming in as.. and what messages your box is sending out.. hope this helps..

Just further to this, a couple of ways that you can setup your VFX number to come in..

If you want to setup an Extension as a DDI you can setup the extension with the correct CLI information..

under Extensions.. click Add Extension.. then select generic sip device.. and setup as follows:

User Extension: Extension you wish to use
Display Name: Up to you

Direct DID: YOURVFXNUMBER - i.e. 9950xxxx
DID Alert Info: YOURVFXNUMBER - i.e. 9950xxxx
Outbound CID: YOURVFXNUMBER - i.e. 9950xxxx

secret: APASSWORD
dtmfmode: rfc2833

You can leave the other settings as default if you wish.. you may want to enable voicemail if you intend to use internal voicemail..

Click Submit

The other thing you can do if you are wanting to setup with an IVR is add in the IVR.. then setup your Incoming Route to send all calls to the IVR..

Click Add Incoming Route, leave all settings as blank except for set destination.. select IVR at the bottom of the page.. submit and apply.. this should send your calls to your IVR.. if you have setup the IVR correctly you can set press 1 for andy.. 2 for candy.. etc etc..

Cheers

Josh

http://wxc.co.nz

mruane

420 posts

Ultimate Geek

#97491 29-Nov-2007 10:34

Tony,

Couldn't help but notice that you had disallow=all in your PEER details after the allow=ulaw&alaw. Isn't that going to effectively remove the ulaw and alaw from the available codecs - because the disallow=all comes after.

EDIT:
PS I did not provide any entry for the USER context or USER details in my Trunnk setup and it seems to be working fine both ways.

Cheers Mike