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9 posts

Wannabe Geek


Topic # 23773 8-Jul-2008 19:49
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Hi All,

I'm wondering if anyone has had any luck configuring the vfx service on other SIP compliant devices.

I've read the "asterisk howto" stuff. But that's specific to an asterisk system. I've tried to get it running using digium switchvox - It won't work. And you can't modify register string etc in switchvox.

Is there any way for vfx to just be "normal" - Perhaps normal enough for me to take my Cisco 7940 IP Phone, and connect it directly to their VoIP service?

I'd really hate to have to buy an FXO and an Linksys ATA and bridge the link in via analog.

Getting information out of callcenter people is not happening. As soon as you mention asterisk - they say. read geekzone goodbye.. and they won't give me any information about their autoprovisioning stuff so I might be able to find out more information.

Cheers

Rob

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  Reply # 144959 8-Jul-2008 20:05
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This has been discussed noodles of times. The VFX configs are locked down and can only be provisioned on certified hardware.
http://www.xnet.co.nz/vfx/hardware.shtml
The alternative is Asterisk, but the helpdesk do not support this; i.e. they supply the SIP trunk, but it's up to you to configure your end on Asterisk correctly. There is a lot of info round here on setting up VFX on Asterisk, and if you post nicely, you'll be sure to get some help too!




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9 posts

Wannabe Geek


  Reply # 146910 11-Jul-2008 17:47
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Hi, thanks for the info.

Would I be right in saying that "Asterisk" is allowed, but not supported. But not just "SIP" - as in no sipX or switchvox or other pabx, just asterisk.

I guess there must be a technical reason that their asterisk SIP trunking stuff has weird things needed to be put into the config file to make it work. I've searched the archives a little bit but can't find any info.

Cheers

Rob

 
 
 
 


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  Reply # 146921 11-Jul-2008 18:10
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Hi Rob
I'm not from WxC, but do work for a WxC reseller, so may be able to answer your question on that one.

WxC only permit authorised devices to be connected to their network, as they want to thoroughly test anything that is being used by their customers, to ensure that all the features of both the devices and the network are fully supported, to ensure that the end user has a good experience with WxC, and to ensure that nothing strange in terms of faults and quirks comes up.

 

Think of it much like the Telepermit system for POTS phones.

Cheers
David




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9 posts

Wannabe Geek


  Reply # 146944 11-Jul-2008 18:40
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Oh well, poor excuse in my opinion. Other voip players out there in NZ aren't so strict and I never have problems with them.

I can't justify having a dedicated asterisk box just to bridge between vfx weirdness and real sip.

I was looking forward to the free 0800 number too. :-)

Thanks for the help :-)

Cheers
Rob


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  Reply # 147054 12-Jul-2008 11:10
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sharkdog: Oh well, poor excuse in my opinion. Other voip players out there in NZ aren't so strict and I never have problems with them.

I can't justify having a dedicated asterisk box just to bridge between vfx weirdness and real sip.

I was looking forward to the free 0800 number too. :-)

Thanks for the help :-)

Cheers
Rob



Hi Sharkdog,
Real SIP !!!, I would love to take the bait here but I won't.....well okay just a bit Wink, this has been discussed numerous times btw but every once and a while I guess I need to put the case forward again.

VFX was designed to be a PSTN replacement service this means we take a great deal of time and effort to make sure that everything works through our whole network, Our network is 100% IP for core switching we only use legacy TDM to carriers that are not yet capable of running IP. Now by being fully IP there are a couple of protocols used , H323 as well as SIP btw this is a very important factor in IP networks.

There's a reason why were the #1 VOIP provider in New Zealand and why we do thing differently to others, Firstly it's because of the Network elements we use, the Network elements are best of breed Carrier grade systems which means they are designed to run high capacity, high volume IN services, others are running open source platforms and whilst they have their places do you think a Carrier would really want to be using this as their core network service offering, how would everyone fell if the whole of the PSTN ran on Asterisk, anyone have any concerns over this ?.

Now moving on why we do only allow certain devices, this goes back to my first comments, our stuff works and I'm not just talking about making a single voice call here, when you say real "Real SIP" we do real SIP and thats all the way through our network. If we control the devices we supply configurations that are known, tested and proved it also allows us to test and prove SIP interworking before any major upgrades, this works because we know the hardware and the underlying SIP stack.

The major problem here when you say real SIP is that and I'm not trying to be rude Sharkdog is that you possibly don't have a full understanding of the Protocol and the impact of deploying it across multiple network elements, The SIP RFC is made up of multiple sub RFC's and different hardware manufacturers implement these in different ways, the net impact of this is that some devices and services may not work together,  i.e Call transfer may not work because one device requires a SIP info, and another may want a SIP 302 or perhaps a reinvite or maybe a SIP refer, these won't all work together even though all methods are SIP standards so our responsibility here is to make everything work that is why we take the effort to test and certify devices.

If we allow any device we would see multiple complaints on things like, I can't use IVR banking services as it doesn't pick up my DTMF , A call I transferred to a mobile didn't hang up and now I have a 12 hour call charge, I can't get my device to register on the network, why doesn't Voicemail Work , now all of these types of problems results in calls to the help desk and the helpdesk people will ask what device do you have ?........now here the problem how may SIP devices are out there and how many SIP stacks do we need to know is it reasonable for all helpdesk staff to know every single SIP device in the world and how it is configured, you would agree with me I presume to say no it is not.

I will actually refer to your case as an example as you would have signed up for Asterisk you actually have all the information required to make this work if you know what you are doing and once again I'm not trying to be rude here, the auto provisioning info you referred to earlier relates only to the devices that we directly support and of no use to you whatsoever. and thats why they work Wink, our set up is a little different but is a purely valid SIP implementation it probably has a little bit more technicality to it than others but thats due a few more security features we use. Helpdesk people are not trained in Asterisk and most probably won't be, there are far to many variables for any helpdesk to be able to support this, it would cost far to much time to be able to offer this as a helpdesk service and we are not a training establishment, people do seem to think that is our responsibility to set it up for them. my hourly rate will be $600 btw Wink, We give you the basic information and advise if you can't configure the device consult the Asterisk community.

Asterisk is not  for basic users it never will be, it's not based around easy use for your standard home user it requires a bit of Technical expertise, our service has not been designed to support everything, other smaller niche VOIP providers do and generally leave you to it and have a lot of issues with services not working, they are not looking to be a mainstream PSTN replacement service we are. This is perhaps why were were TUANZ Carrier of the Year last year (First time a Tier II has been awarded this) and also chosen to supply VOIP services for the pilot Telecom FTTH project Cool.

We will add more and more devices going forward and Cisco IP phones are ones that are capable of being used, so you may want to watch this space as we add more and more offerings.

Sorry if I come off sounding a bit over the Top but the Real SIP , poor excuse comment probably provoked me just a little bit Tongue outand I hope you see where we are now coming from.



Regards
Phil Moore
Operations Manager
WorldxChange Communications




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

             

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9 posts

Wannabe Geek


  Reply # 147229 12-Jul-2008 20:48
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Hey,

Thanks for a very in depth reply! -

I'm reasonably aware of how wildly different each device manufacturer tries to implement their "real SIP". I wouldn't think of myself as a guru tho. In fact I have deliberatly tried to stay out of the fine technical details because I'd rather just find a system that just works out of the box. Thats why the enquiry about switchvox (actually runs asterisk behind the scenes).

I guess using switchvox won't be an option. It only has a few options for a SIP providor. authname, authpassword, server/proxy. Plus a few things about how it handles dtmf etc. With no possibility to have a customer register string, I guess i'm out of luck.

Having selected authorized devices certainly is one way of ensuring that everyone gets a good experience. I can understand how it wouldn't make sense to support the minority. the vfx service is clearly marketed as a residential phone replacement anyway, Heres me hoping that the vfx line that comes bundled with my adsl could be usefull as another "free" DDI going into my switchvox system.

to give you a background on my project; I'm providing hosted pabx services. Each company gets their own switchvox either on VMware or bare metal in the datacenter, or a box on their site whatever seems best for their needs. It takes under 10 minutes to provision a fully working switchvox 100% ready to go, and I can let the client into the admin area without worrying about them breaking stuff, It's very simple to use. It takes even less time to provision by duplicating a VM.

Thanks for explaining a bit about your POV, it's helpfull, and at least now I know I either need to spend the time and use/learn pure asterisk, or give up and use another provider without the extra "security"

Cheers

Rob

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  Reply # 147231 12-Jul-2008 20:57
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Switchvox should actually work fine since it is just Asterisk with a nice GUI..



9 posts

Wannabe Geek


  Reply # 147232 12-Jul-2008 21:03
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I'd give you a box of beer if your magic fingers could enter the information more accuratly than mine :-)

Trust me, Where it asks for username i put it in, where it asks for password i put that in, and so forth. Also since you can't get into the hosts file to add a static entry for the as.wxcnz.net host i had to add a static entry in my dns cache server so it would at least see the server.

Whats with not having an A record? thats kinda weird.. It's not that hard to set up an A record. :P

Unfortunatly the trouble and the beauty of switchvox is that it only has certain boxes... put the information in there... if it doesn't work... then you pretty much screwed, there is no console, there is no editor for directly editing config files.

Cheers

Rob

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  Reply # 147236 12-Jul-2008 21:15
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I've never actually played with it as I've always considered it to be quite an unusual product!

What exactly appeals about the product over other options in the maketplace? Trying to run any VoIP option on a virtual server is also asking for trouble.




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Wannabe Geek


  Reply # 147238 12-Jul-2008 21:24
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Well, its not open source. Thats one thing. :-(

Whats nice about it is that its very simple. You just install it, it has a quick wizzard (like 3 or 4 screens worth) to configure its IP. then you point your browser at it. It works with every phone I've tried, and works well. In under 10 minutes I had it set up under vmware esx with 2 phones hooked up and a 2talk connection set up and was starting to play with IVRs already!

It doesn't do anything like phone autoprovisioning. Which would be cool, but what it does do it does well. Very simple dialplan builder. Very easy to allow/disallow different dialplans to different extensions a monkey could configure it. Even the ever well meaning IT guy at the clients company who tends to break more stuff than he fixes could log in and add extensions / voip providers.

Also phones just seemed to work.. I had trouble getting my phones to work with trixbox. Other people have told me they never had troubles with trixbox but i seemed to. following all the howtos word for word.

I've got a few 20 user companies running on vmware esx i never have any problems. probably 10 concurrent calls across the lot at once, so not high load. Have yet to run sipp over it.

Cheers

Rob

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  Reply # 147310 13-Jul-2008 09:18
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sharkdog:

Trust me, Where it asks for username i put it in, where it asks for password i put that in, and so forth. Also since you can't get into the hosts file to add a static entry for the as.wxcnz.net host i had to add a static entry in my dns cache server so it would at least see the server.

Whats with not having an A record? thats kinda weird.. It's not that hard to set up an A record. :P


Cheers

Rob


Hi Rob

Actually this is one the SIP interop issues that I was explaining and incorrect implemantation of SIP that we see in a lot of generic SIP offerings, when you use an OBP (Session Border Controller) then the proxy information is suppossed to be sent in the SIP messages it does not require DNS lookup for as.wxcnz.net before it sends the message as the SBC can resolve it the Endpoint does not , thats what OBP is for. You do not want these network elements exposed in anyway to the real world, to many Black Hats out there Smile, not using any from of SBC is a pretty dangerous deployment.


Just took a quick look at the admin quide for this box and it appears it does not support Outbound Proxy at all which will cause problems, as a quick test, try setting your proxy to 58.28.20.150 and see what you get from that.




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  Reply # 147313 13-Jul-2008 09:31
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sharkdog: Hey,

Thanks for a very in depth reply! -



Thanks for explaining a bit about your POV, it's helpfull, and at least now I know I either need to spend the time and use/learn pure asterisk, or give up and use another provider without the extra "security"

Cheers

Rob



Just a note Rob, IP security is going to be a big issue and protecting the customers is a big deal and is worth spending a little extra time on, SIP fraud will be  be on the rsie and there have been known case's already with other VOIP Providers. So isn't it good to know that we do take it very seriously and we have the customers security looked after. 




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

             

https://www.facebook.com/wxccommunications



9 posts

Wannabe Geek


  Reply # 147363 13-Jul-2008 12:08
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The extent of the switchvox debug logs is:

Your VOIP provider, 58.28.20.150, is not responding to SIP OPTIONS requests. This may indicate that their SIP service is not functioning properly. Please contact your VOIP provider.

 1  07/13/2008 12:07 PM  WARNING  chan_sip.c: Got 404 Not found on SIP register to service [bob]@sip_provider_108, giving up


Cheers

Rob

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