Avaya IPO and Remote Phones 2

[chazrc]

Can someone provide a list of ports that need to be forwarded to get the 1616 phone working remotely? I am on the new 8.0.16 release. Phone says Discover and the Public IP on the screen.
Also, I can see the phone trying to login or lat least mention of that extension in monitor, but no luck yet.

Ping is on on the router that the IPO is on, and we set the h.323 extension up, and allowed remote extensions in the user profile.

Thanks,

Chaz

 

[amriddle01]

Does

 

[bonus99]

Thanks for your comments.

I changed 1720 to both UDP and TCP but no difference.

So near yet so far...


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I don't think I know anything.
I am never able to help anyone!

 

[amriddle01]

I think you need to run STUN with a public STUN server, see what firewall type it detects, it may never work if it's one of the unsupported types

 

[qtelcom]

No voice means blocked ports/misdirected traffic normally"

Since we are on this subject, I thought I'd post my issue.

I have my IP Office behind my office router, and also a remote phone (9650) behind the home router. The remote phone can register with no issues.

I have the issue of no voice, I can see on my office router that it is indeed blocking the udp ports, however, the ports it's blocking are not the ones I'm using in the RTP Port Number Range (Remote Extn) which I have them set to 51102-53000. This range is correctly forwarded to the IPO Lan1 IP=10.0.0.4), as are the other ports listed in Manager (1718,1719,1720). A correct default route in Manager is programmed to my router (0.0.0.0/0.0.0.0/10.0.0.202/LAN1)

for example, I make a voice call to my remote phone and I can see udp traffic on port 51152 as my source out to the remote phone, but the remote phone is trying to use udp ports outside of my 51102-53000 range, and thus gets dropped by my router. What controls or determines what upd ports the remote phone uses? The 9650? the home router? Why is it picking random udp ports outside the range I specified in manager?

 

[bonus99]

I ran STUN (69.90.168.13) and it reports blocking firewall and no IP set.

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I don't think I know anything.
I am never able to help anyone!

 

[amriddle01]

Will never work with Blocking firewall

 

[bonus99]

This may be a really stupid question but...

How do I get round that?

I have a couple of sites where I am trying to get this working and both have come up with blocking firewall.

Sorry if I am ignorant to the obvious

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I don't think I know anything.
I am never able to help anyone!

 

[amriddle01]

Good question, to which I don't know the answer, I would say you need to change the way the router/firewall works or change the router/firewall.

The docs say:

If STUN reports the Firewall/NAT Type as one of the following, the network must be reconfigured if possible as these types are not supported for remote H323 extensions: Static Port Block, Symmetric NAT or Open Internet

It doesn't mention "blocking firewall" specifically but I would say it isn't good

 

[bonus99]

I read that and thought the same.

One site just has a Linksys router and the other site has a zywall gateway, we just put the zywall in and it is a good piece of kit.

I feel a RANT coming on...

Not sure how this can be so difficult!

Thanks for your help, I will plod on and let you know how it goes.

~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
I don't think I know anything.
I am never able to help anyone!

 

[amriddle01]

You can rant, but rant at the firewall, this is stuff outside the control of the system/handset, it can't force it's way through firewalls otherwise there would be no value to owning a firewall

 

[bonus99]

You are quite right, I am just frustrated.

It would be useful for AVAYA to do a bit more in their documentation though.

Like, how to overcome certain issues rather than just saying, "you need to see this"

I appreciate your help and hope to report back positively very soon



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I don't think I know anything.
I am never able to help anyone!

 

[NLPeterK]

Avaya cannot document sth that is out of their control

Just for humour, try a different stun server.. the one that's in there by default doesn't always work in my experience... Try 132.177.123.13.

Are you running stun on the right interface btw? I understand you have your sip trunk running on LAN2 and this whole remote phone scenario on LAN1?

 

[bonus99]

Hi Peter - thanks for that. I tried it and it has reported back FULL CONE NAT and also populated the public port as 5060.

The phones connect OK, they can dial - hooray but...

Only one way transmission, when dialling in or dialling out, the IP OFFICE phone can hear the other end, however, the remote caller cannot hear me.

I will do some digging

Cheers

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I don't think I know anything.
I am never able to help anyone!

 

[NLPeterK]

Are you sure you have no conflicting ip routes?

Just for fun, can you post all the ip routes in your system? Also describe what you think you need them for.. Maybe we can clean them up or fix them

 

[bonus99]

ok then, just for fun...

I have 3 x routes

0.0.0.0 0.0.0.0 10.1.5.1 LAN1 - this is for route all
xxx.xxx.xxx.xxx 255.255.255.255 xxx.xxx.xxx.xxx LAN2 - this is for the SIP trunk
192.168.99.0 255.255.255.0 0.0.0.0 - Remote Manager

Over to you

~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
I don't think I know anything.
I am never able to help anyone!

 

[NLPeterK]

Are you calling out on the sip trunk by any chance?

I see you have created a route for the sip trunk which consists of a single adress (255.255.255.255). That is hardly ever the case. There is a sip server on adress x, but RTP (=audio) often streams to and from other adresses, probably in the same range.

That would explain the lack of audio here and there.

How's the audio when you call an internal extension?

 

[bonus99]

There are no internal extensions, it is being set up specifically for remote users. My colleague has called from there (via SIP, the only trunk)to an analogue line and it was fine.

I get your point with the SIP and there is a /29 subnet so I have set the mask at 255.255.255.248

AND IT WORKS!

YOU ARE A STAR AND I WILL AWARD YOU ONE!

I will do some testing and take out a lot of config until I get to the point where there is enough for it to work and post a final report.

Thanks all for your help on this

Cheers



~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
I don't think I know anything.
I am never able to help anyone!

 

[NLPeterK]

Yay! My pleasure!

 

[bonus99]

Here is an update...

I have 2 x 9600 and a 5602 SW working from one satellite site to the main site.
The 5602 works OK, not the sound quality of the 9600 though but does work.

Problem though is with the other satellite site, phones register OK and I can ring them, they answer and we can talk.

However, when they call me, the phone rings, I answer and there is no audio either way. This site is in Madrid, not that it should make any difference?

Any ideas greatly investigated.

many thanks



~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
I don't think I know anything.
I am never able to help anyone!

 

[brainey1]

I had the same issue with the far end using ATT Uverse. Finally found out that when a call went out on the uverse, it was using the gateway address of the uverse modem which was different than the static ip program on the firewall I was given from ATT. So the audio didn't know how to get back to the uverse side. Called ATT and they said that it was functioning correctly and they couldn't make it send from the static ip on the modem.