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Discussion Starter #1
I'm thinking of picking up a few Aastra 9133i, and hoping to register them directly to voip.ms (i.e. without any onsite PBX).

Anyone have experience with this? Any problems?

An OLD review of the 9133i suggested this wasn't possible, but I don't think that is accurate anymore.

Yoder.
 

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I'm thinking of picking up a few Aastra 9133i, and hoping to register them directly to voip.ms (i.e. without any onsite PBX).

Anyone have experience with this? Any problems?

An OLD review of the 9133i suggested this wasn't possible, but I don't think that is accurate anymore.

Yoder.
I use four Aastra 9112i in my house, all directly connected to the network without an ATA or PBX. I love it, and it works flawlessly when i configure them all on sub accounts.

An added benefit of this approach is that i can call room-to-room, and can also have as many incoming or outgoing lines as i need. It's not uncommon when the kids are home from college to have three or more parallel calls going on at the same time.
 

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I own myself an Aastra 9143i and I love it. It supports up to 9 lines (3 hard keys and 6 other soft buttons which can be configured as additional lines. I have Lines 1 and 2 setup for voip.ms, line 3 is the freephonelinne.ca while one of my soft buttons (configured as line 4) is used for "Freephonie", a French ISP which gives me free calling to France. It works just like an office phone. If you are on a call and someone calls you, it will alert you and ring on line 2. You can conference in, place calls on hold (with music on-hold supported by VOIP.MS). It's an amazing phone. If you have multiple units, like 99semaj, you can do extension calling, transfers, etc..

Also, the display is great and the phone looks very professional on your desk. You'll love it!
 

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Discussion Starter #4
thanx for the info you guys!

I'm going to try them at home first, but anticipate installing them out my small business office.

Because of a peculiarity of my setup, the internet is provided by a third party, and I don't have ANY control over the router... I can open up a ticket with them, but won't be able to trouble-shoot the router LIVE.

I'm hoping it is just "plug-n-play"...
 

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Discussion Starter #6
Aastra 9113i incoming call problem solved with "registration renewal timer"

Just in case it is useful to anyone.

I tried setting up 2 Aastra 9133i phones on my network. Registered fine, sent outgoing calls fine, but incoming calls were unreliable (they would work for about 1 minute after 'restarting' the phone, but then wouldn't come in), and my ring group didn't work.

I had the NAT option enabled on the voip.ms side of things.

I fooled around with a variety of things and ultimately this is what worked.

In the Advanced SIP settings of each phone there is an option called: "registration renewal timer" which is described in the admin manual as: "the length of time in seconds prior to expiration that the phone renews registration". The default was set at 15.

Once I set it to 60 seconds all of sudden my phones would transiently be able to receive incoming calls for the 60 seconds prior to the next voip.ms registration period (as found on the voip.ms main portal page). I noticed the registration intervals on that page were 2 minutes to 2:45m, and eventually I changed the "regisration renewal timer" to 180 seconds (3 minutes), and all of a sudden my phones worked fine, received incoming calls always, and suddenly my ring group started working fine as well.

I'm wondering if the "registration renewal timer" is the equivalent of "NAT Keep Alive"?

Any thoughts?

Yoder
 

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Hi Yoder,

I just checked my 9143i info and the registration period is set at 60.
In the advanced SIP settings, Registration Failed Retry Timer is set at 1800, Registration Timeout Retry Timer is 120 while Registration Renewal Timer is 15. It's been a while since I configured it, but things are pretty good, so I don't touch it.

Let me know if you have any questions about specific settings and I can let you know what I use.

Just out of curiousity, did you get a good price on your 9133s?
 

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Discussion Starter #10
NAT keep alive on Aastra?

I'd be happy to tell you info - he only listed 4 for sale.

PM me if you want the contact info...

BTW: do you know if these Aastra phones have a "NAT keep alive" function - or equivalent?
 

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I'd also be interested in knowing where I can pick up a unit. I need to get a new office phone setup in the next week at the house for a new job.

Could you PM me some details?
 

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Actually disregard above. I found a 6757i today that I purchased. However I am having a HECK of a time getting it to work with Voip.ms. It does however work with a FPL account that I have. As a side note, I have been able to get my voip.ms account working with Groundwire on my iPhone 4.

Are there any tricks to getting this 6757i working with Voip.ms ? I've tried so many different ways this afternoon that I'm going cross-eyed and need to take a nap :p

If anyone can assist, please let me know what info you need to get things going? All I want to do is register it directly, there is nothing between me and voip.ms other than my router.
 

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Discussion Starter #15
gramzee - just send us your info on your phones (minus the password)

remember, in the basic SIP authentication settings, the "phone number" actually has to be your voip.ms username, or that of a sub-account you've set up. Not the phone number.
 

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Discussion Starter #16
registration vs. registration renewal timer

OK so here's my interesting dilemma.

I have 3 phones setup each with its own subaccount.

I have registration set at 60, and registration renewal timer at 15. It only dials 1 or 2 of my phones (and occasionally none) when receiving an incoming call.

To reliably fix this I change my registration renewal timer to > 60 (i.e. more than the registration).

According to the manual the registartion should always be > than the renewal timer.
Doesn't make sense - but is working for me?

(also tried turning NAT on/off/route on the voip.ms - but no reliable change)

Yoder
 

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HAH! Looks like that was the problem. Not sure if I have all my other settings tweaked. How do I get you the info? Screenshots would be too big, is there a way to export the settings? Or just type it in manually?
 

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Ok this is what I have so far. Seems to only export these lines to the local.cfg file. Everything else in the phone is set to defaults. It's dialing out, haven't tested call quality, kind of late to bother anyone.

sip line1 auth name: 121075
sip line1 password: xxx
sip line1 user name: 121075
sip line1 display name: 16474786395
sip line1 proxy ip: toronto.voip.ms
sip line1 proxy port: 5060
sip line1 outbound proxy: toronto.voip.ms
sip line1 outbound proxy port: 5060
sip line1 dtmf method: 0
sip line4 auth name: 12896274321
sip line4 password: xxx
sip line4 user name: 12896274321
sip line4 display name: 12896274321
sip line4 proxy ip: voip.*************.ca
sip line4 proxy port: 5060
sip line4 outbound proxy: voip.*************.ca
sip line4 outbound proxy port: 5060
sip line4 dtmf method: 0
 

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OK .... sorry for the posts, but i'm up late playing :)

I can now dial out on the phone, but am not receiving calls. I also subsequently upgraded the firmware, as what was on the phone was quite old compared to what i've been reading on various sites. So now, likely things will be in line with whatever tips people might be able to provide.

If there is any info needed to help troubleshoot that wasn't in the log I posted above, please ask and I'll forward it along.
 

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Dialing out suggests that the phone is originating calls ok, and therefore your setting are correct.

If calls are not terminating properly, perhaps it's your web portal setting that are wrong. Have you set up a ring group for your DID?
 
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