57i Not Registering - No Service

Zion800
Posts: 76
Member Since:
2007-03-06

Hi, I know this has been discussed on the forum, however none of solutions discussed worked for me. I have an Aastra 57i that is not registering to my Trixbox. Here is some information on my setup:

- Trixbox 2.6.0.7
- Aastra 57i (firmware 2.2.1)
- Trixbox is NOT behind a NAT router and has its own public IP.
- Firewall ports 5060-5082 (UDP), 10000-20000 (UDP) are open
- 57i is located on another network, and is behind a NAT router.
- I used the Trixbox endpoint manager to configure the phone, which worked using TFTP from my remote location to Trixbox

The odd thing is that the phone can place calls, and they were perfection and the audio is great, however, the phone cannot receive calls. When I look at Trixbox, the phone has not registered.

Any help would be greatly appreciated.

-Michael



KodaK
Posts: 1885
Member Since:
2006-06-14
Have you forwarded any ports

Have you forwarded any ports to the 57i on the other network?

--

WARNING: I no longer actively participate in these forums. My thoughts on trixbox in a nutshell: http://www.youtube.com/watch?v=q4xBMkWu1pE Use AsteriskNOW instead.



Zion800
Posts: 76
Member Since:
2007-03-06
No, however, I don't see why

No, however, I don't see why I need to. I have a Linksys SPA-942 that works fine on the same network as the Aastra 57i.

-Michael



KodaK
Posts: 1885
Member Since:
2006-06-14
Quote: No, however, I don't
Quote:
No, however, I don't see why I need to.

Because without a proxy or some port forwarding you're going to have problems with more than one phone behind a NAT device.

--

WARNING: I no longer actively participate in these forums. My thoughts on trixbox in a nutshell: http://www.youtube.com/watch?v=q4xBMkWu1pE Use AsteriskNOW instead.



Zion800
Posts: 76
Member Since:
2007-03-06
Gotcha. So how would this

Gotcha. So how would this all work if I wanted 3 Aastra 57i phones on this network?



Zion800
Posts: 76
Member Since:
2007-03-06
I just forwarded the

I just forwarded the necessary ports directly to the Aastra 57i, and no luck. I think this No Service problem is something else...I entered a STUN server in the settings and tried that as well, and it still didn't work.

-Michael



KodaK
Posts: 1885
Member Since:
2006-06-14
If your router at the remote

If your router at the remote end supports UPnP then you can try enabling that on the Aastra phones. Try that first, it's quick and easy.

Past that, I'd need to know a lot more about your network(s) and the resources available to you.

One way is to use a different SIP communication port for every phone, but that gets pretty messy real quick. Another is to use a SIP proxy like SER (google openser.) You can also use a VPN tunnel between the two networks so that NAT is not an issue, but this introduces VPN overhead and there are QoS hurdles to overcome when VoIP traffic is tunneled.

Try the UPnP thing first.

--

WARNING: I no longer actively participate in these forums. My thoughts on trixbox in a nutshell: http://www.youtube.com/watch?v=q4xBMkWu1pE Use AsteriskNOW instead.



Zion800
Posts: 76
Member Since:
2007-03-06
Ok, so I decided to connect

Ok, so I decided to connect the phone directly to the cable modem so its no longer behind NAT and it registered with Trixbox, which means you were right on the money! However, before I did this, I enabled UPNP on my router and on the Aastra 57i. As it rebooted, it did some UPNP Port Mapping, but the phone did not register.



SkykingOH
Posts: 8081
Member Since:
2007-12-17
DD-WRT voice also has a SIP

DD-WRT voice also has a SIP proxy. You can register multiple phones behind it. Many routers support this open source firmware. Take a look at www.dd-wrt.com

This firmware also supports simple QoS.

If these phones are going to be heavily used I would consider placing another server at the location of the Aastra's. You can then trunk using IAX2 which has many advantages over SIP.

The hardware to run 3 extensions is so minimal. I run 7 phones at my house on a PIII 800Mhz/512MB ram and I have no problems at all.

Scott

--

Scott

aka "Skyking"



Zion800
Posts: 76
Member Since:
2007-03-06
Its really not a heavily

Its really not a heavily used location. Is there no way I can get this to work?



SkykingOH
Posts: 8081
Member Since:
2007-12-17
Did you look at the DD-WRT

Did you look at the DD-WRT solution? Pick up a compatible router ($50) download the software, set the phones to register to the router (it's a simple proxy) and you will be up and running without any hassles.

--

Scott

aka "Skyking"



KodaK
Posts: 1885
Member Since:
2006-06-14
Um, how do you get "no way"

Um, how do you get "no way" out of the half dozen or so options we've presented to you so far?

Yes, all of the options will take some work and/or equipment, but it can be done.

--

WARNING: I no longer actively participate in these forums. My thoughts on trixbox in a nutshell: http://www.youtube.com/watch?v=q4xBMkWu1pE Use AsteriskNOW instead.



Zion800
Posts: 76
Member Since:
2007-03-06
Please excuse me, I meant

Please excuse me, I meant ask if there was any way to do this without purchasing additional hardware. See, my gripe here is that I can have a Linksys phone and Grandstream adapter simultaneously working perfectly fine on this network, but when it comes to setting up the Aastra, it doesn't register (but can place calls). I feel like this is more of a configuration problem and something that can be fixed without purchasing additional hardware. I am the type of person who is willing to work through a headache and fix a problem if there is any reason for me to believe there is a possibility for it to work. Since I already have two SIP devices behind this network, and they work flawlessly, I believe there is a possibility to fix the Aastra without purchasing additional hardware.



SkykingOH
Posts: 8081
Member Since:
2007-12-17
I guess it's a matter of

I guess it's a matter of perception of value of time. Sure roll up your sleeves turn on the SIP debug on the remote end and start changing Aastra values until it works. SIP and NAT has known issues, that is the reason for proxies, Session Border Controllers and all the other solutions that have been offered to support SIP NAT traversal.

Since I have never tried to do this I can't help you with specifics. The time invested to fix it is worth the $50.00 IMHO.

The proxy scales well, you can see all the devices registered to it and it works without confusing configurations. Since DD-WRT is a Linux build you can run add the DHCP option 66 and run a local tftp server. I would never consider exposing tftp on the Internet.

The initial configuration of DD-WRT is all web based so it is very quick to deploy.

I am a huge advocate of this software for small installs.

Scott

--

Scott

aka "Skyking"



KodaK
Posts: 1885
Member Since:
2006-06-14
Here's what you're going to

Here's what you're going to have to do:

1) turn on SIP debugging to see if any registration attempts are even being seen by the * server.

2) sniff the remote network to see what's happening on that end, as far as registration goes.

Any further steps depend on what happens with those. Be prepared for a long, drawn out process.

--

WARNING: I no longer actively participate in these forums. My thoughts on trixbox in a nutshell: http://www.youtube.com/watch?v=q4xBMkWu1pE Use AsteriskNOW instead.



xinsight
Posts: 18
Member Since:
2008-01-17
i'm a little late to the

i'm a little late to the discussion - did you get the phones working?

I think the original poster's approach to understanding a problem before simply buying more hardware is reasonable.

I've found that port forwarding is only useful if you have a single device behind the firewall. Since you have multiple phones, I would turn all forwarding, upnp, DMZ, etc. off. The router will figure out how to route packets correctly. That's basically what it's designed to do. ;)

The problem with NAT routers is that they all have slightly different behavior. Asterisk has lots of config options for nat, but i've found the key ones for SIP are:

nat=yes
canreinvite=no
qualify=yes

The qualify line keeps the NAT connection open - i believe my sending the phones a meaningles OPTIONS request. So if you do a 'sip show registry' from the asterisk console, you will see your sip phones and a ping value in millisecs.

The usual problem with routers and NAT is one-way audio - you can't hear the other person or they can't hear you. Registering is usually not much of a problem.

Hope that helps.

--

Build your own PBX
pbxer.com



Zion800
Posts: 76
Member Since:
2007-03-06
Thanks xinsight, I was able

Thanks xinsight, I was able to get everything working just as you mentioned without buying new hardware ;p



jahyde
Posts: 2002
Member Since:
2006-06-02
so how did you do it?

so how did you do it?

--

--my PBX is run on 2 V8's



Zion800
Posts: 76
Member Since:
2007-03-06
I believe xinsight posted

I believe xinsight posted the options that need to be set in order to get things working...let me know if you need more clarification...



grandanet
Posts: 5
Member Since:
2010-01-03
Solution for "No Service"

This is a registration timeout message, the solution is to set the SIP registration times to never expire.

For example, Aastra 480i

Go to -> Global SIP

Under the = Basic SIP Network Settings

Registration Period = 0

Under the = Advanced SIP Settings

Registration Failed Retry Timer = 1
Registration Timeout Retry Timer = 120
Registration Renewal Timer = 0

Let me know if this helps...

--

--------------------------
GrAnDaNET :-)



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