TIPS Nortel IP 1535 $27.95

w1ve

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As long as the DID is SIP reachable, you go to e164.org and register it.
 

mtennant

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Very confusing sign up process. I wasn't sure what to enter for the SIP://____________ entry. Anyway, I got registered.
 

wardmundy

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[email protected] should do it at e164.org using your sip2sip number. Then you can add an IPkall DID to the same SIP address. And finally you can get a local Google Voice number, point it to your IPkall number, and your Nortel phone will be accessible to everybody on this planet anyway...

StairwayToHeaven-D-4d.jpg
 

mtennant

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I've been around VOIP for several years, but doing all that just seems a bit much and confusing. I think I'll pass on this particular stairway to heaven.
 

mtennant

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Because you say so Ward, I will try it! The only thing I don't have already is the IPKall number.
 

wardmundy

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Just be advised that thousands of folks (lots of PIAF users) have done this and Google Voice doesn't particularly like IPkall because of high tariffs on the inbound calls. IPkall is a CLEC and they get $'s for each call made to the number.

If you don't use an IPkall number for a month, it gets disconnected. But many never cleared them out of their GV accounts. So... you may get a message from GV when you attempt to register the IPkall number that it's already registered somewhere else. If this happens, about all you can do is cancel the IPkall number and try again. I've had good luck with IPkall's 253 area code. Nobody seems to like that one. :wink5:
 

mtennant

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When I try to point my GV number to the IPKall number, I get this error.

"This number is already being used on two other Google Voice accounts. Forwarding numbers cannot be shared across more than two accounts."

I can only imagine that the IPKall number has been around the block a few times, used by people with GV numbers, not used, and taken away by IPKall, only to be given to me, who now cannot use it.
 

rxcomm

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I'm still trying to trunk SIP2SIP with PIAF, so I can video into the PBX from a SIP URI. It works for 1 or two calls, then I immediately get congestion on further calls. No Video works.
Bummer.
I don't think sip2sip.info likes the h264 codec. You have to force h263 on them both inbound and outbound. When I do that, video works both inbound and outbound through PIAF if I initiate a video call by pressing the "video camera" button to dial.

However, I can't transition from an audio call (initiated by pushing the green button) to a video call through PIAF (by pushing the "video camera" button) after the call is initiated. Evidently this is a bug in Asterisk? Perhaps arising from the video codec negotiation bug? I'm at Asterisk 1.4.35.

Dave
 

w1ve

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I don't think sip2sip.info likes the h264 codec. You have to force h263 on them both inbound and outbound. When I do that, video works both inbound and outbound through PIAF if I initiate a video call by pressing the "video camera" button to dial.

However, I can't transition from an audio call (initiated by pushing the green button) to a video call through PIAF (by pushing the "video camera" button) after the call is initiated. Evidently this is a bug in Asterisk? Perhaps arising from the video codec negotiation bug? I'm at Asterisk 1.4.35.

Dave

Hi Dave,

Can you post your trunk/extension settings?

I'm using h263, and it still does not work reliably (no video at all). I can make 1 or two calls, then I always get congestion.
I'm just using allow=ulaw&h263 on trunk and on extension.

My suggestion on video after audio? try setting reinvite=yes on both the trunk and the extension. The video RTP needs to be renegotiated when you press the video button, and the way to do that is by having the endpoints send another SIP INVITE message during the call -- a RE-INVITE -- to set up the RTP for video. Right now, I'd just like to get the trunk working.

Gerry
 

w1ve

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Ahhh, the price of FREE. :rolleyes:

Get a voip.ms/callcentric DID for $1.49/month.. That's $17.88 a year for a ROCK solid DID. a SIP URI ([email protected], for example), SIP dialing comes free from both companies, and no per-minute.

Register the DID with e164.org and rock on.

I want my cellphone reachable from SIP; I registered the number with e164.org, pointing it at my PIAF. Then, I simply redirect the incoming call to my cell via an ITSP. Yep, I pay a little bit for the final leg, but now I have a world-reachable cell phone.

Rather than totally free -- which usually has some significant compromises -- smart application of some of the free tech + PIAF will result in some very flexible, very inexpensive phone service.
 

rxcomm

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sip2sip configuration

Here are my settings. I will change the reinvite as you suggested above tonight and see if that works. Note that 2233xxxxxx refers to the sip2sip id and 2233XXXXXX is the dialplan.

Here are my trunk settings:

Trunk name:
sip2sip

Peer details:
username=2233xxxxxx
type=peer
secret=xxx <-this is the sip2sip password
nat=yes
insecure=very
incominglimit=1
host=sip2sip.info
fromuser=2233xxxxxx
fromdomain=sip2sip.info
disallow=all
context=from-trunk
canreinvite=no
caninvite=no
allow=g729&h263

No user context or user details.

Register string:
2233xxxxxx:[email protected]/2233xxxxxx

Here are my outbound route settings:
Dial pattern:
2233XXXXXX

Trunk sequence:
SIP/sip2sip

Here are my inbound route settings:
Description:
sip2sip.info

DID number:
2233xxxxxx

Destination:
Extension 710

Here are my settings for extension 710
secret: xxx <-this is the extension password
dtmfmode: rfc2833
canreinvite: no
context: from-internal
host: dynamic
type: friend
nat: yes
port: 5060
qualify: yes
callgroup:
pickupgroup:
disallow: all
allow: g729&h263
dial: SIP/710
accountcode:
mailbox: 710@device
deny: 0.0.0.0/0.0.0.0
permit: 10.8.0.0/255.255.255.0

I also added the following to sip_custom.conf to force the inbound codec to h263 - will fix this when I have a minute to work on it (connect inbound route to sip2sip trunk).

sip_custom.conf:
disallow=all
allow=g729
allow=gsm
allow=ulaw
allow=h263

I'm a rookie at this, so if somebody's got a better suggestion, let me know.

Dave
 

tshif

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These phones are interoperable with the Grandstream GXV 3000.

The Grandstreams perform better when using h264 - the interop is reliable and seems solid.

This is the first video phone I have found to interoperate this well with the Grandstream GXV3000.

For what you might pay for a license for a video enabled softphone (if there were any left that worked with our setups) - these units seem to be a great bargain.

My units arrived already setup for english, they are Hardware version 1.0, running firmware 0.2.50.

Wifi works GREAT - and performance of video and audio is identical to wired lan. The Wireless LAN its on uses only MAC address registration - so I havent had time to experiment with the various wireless security options.

Thanks for the tip that these were available.
 

wardmundy

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When Nortel released this phone, my recollection is that it sold in the $700-$800 price range. So, yes, it's a pretty good phone... probably the best one Nortel ever made. Now that we've cleaned out all of the turkeys, the newer U.S. firmware seems to be an improvement. WiFi security is flawless.

For video codecs, choose H.263. Then deselect H.264 in first place. Then replace H.263 in second slot with H.264. Then go back to first place and choose H.263.

Works great as an Asterisk extension, too. We're taking a few to the mountains today to play this week.

nortel-1535.jpg
nortel-1535.jpg
 

kenn10

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Has anyone figured out how much bandwidth these things use in each direction when showing video?

At my NC mountain house, the best DSL I can get is 2mb by 768k. Wondering how they'd perform back to my main system from there.
 

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