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Unread 04-05-2009, 01:03 PM   #1
akhater
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Default E71 behind DSL router TG585V7 issues

I'll try to summarize my issue

I have 2 voxalot accounts one registered on the PC and one on the e71 on DSL behind a TG585 V7 router

1) I can register on both eyeBeam and e71
2) I can call from e71 to eyeBeam (voice is fine both ways)
3) I CANNOT call from eyeBeam to e71 i'm redirected to the voicemail
4) I also tried calling from another internet connection to the e71 behind the DSL router same thing not working
5) Finally from the e71 I can only call voxalot members, if i try to call someone at callcentric it directly gives "internet service not available"

can anyone help with these issues please

thanks

Last edited by akhater; 04-05-2009 at 04:12 PM.
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Unread 04-05-2009, 02:24 PM   #2
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What do you mean by:
Quote:
2) I can call from e71 to voxalot (voice is fine both ways)
and in the same time
Quote:
3) I CANNOT call from e71 to eyeBeam i'm redirected to the voicemail
Did you mean you can call Voxalot's echo test at *600?
Quote:
4) I also tried calling from another connection to the e71 behind the DSL router same thing not working
How exactly did you try to do that?
Quote:
5) Finally from the e71 I can only call voxalot members, if i try to call someone at callcentric it directly gives "internet service not available"
Are you using SIPBroker to dial or direct URI? What syntax you're using for this purpose?
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Unread 04-05-2009, 04:09 PM   #3
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Quote:
What do you mean by:
2) I can call from e71 to voxalot (voice is fine both ways)

and in the same time
3) I CANNOT call from e71 to eyeBeam i'm redirected to the voicemail
I am sorry this is a mistake, I CAN from e71 to eyebeam (all is fine) but from eyebeam to e71 I'm redirected to voicemail as if the e71 was not registered

Quote:
Did you mean you can call Voxalot's echo test at *600?
Good question i've just tested it 600 and 500 work but i CANNOT hear, from e71 to eyebeam audio IS fine

Quote:
4) I also tried calling from another connection to the e71 behind the DSL router same thing not working
How exactly did you try to do that?
A friend of mine has voxalot account he tried calling me on the e71 and it didn't work (voicemail), he tried calling me on eyeBeam it worked.

Quote:
5) Finally from the e71 I can only call voxalot members, if i try to call someone at callcentric it directly gives "internet service not available"
Are you using SIPBroker to dial or direct URI? What syntax you're using for this purpose?
from eyebeam i just call number@callcentric.dom the same thing does't work from the e71, no SIPBroker, I also tried using the iNUM number same thing
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Unread 04-06-2009, 10:37 AM   #4
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It could be that you've mistakenly placed your E71 softphone in DND mode. I often see people do that without knowing so by dialing certain codes or enabling it from their ATAs. In that regard check that your E71 isn't placed on DND in which case incoming calls are routed to your voice mail as you describe.

Second, it could be codec mishandling. It's possible the voice codecs on your E71 to fail to communicate with the server or end clients.

Trying calling different echo tests on different severs, for example:
*266301
*645904
*850301
See if you get any results.

Lastly, because you're behind a router the NAT handling with Eyebeam may not be as good as your E71. Perhaps you should post your settings so we can go from there. As last resort, you'll have to enter the web configuration of your DSL router and do some port forwarding.
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Unread 04-06-2009, 03:51 PM   #5
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The only reliable way to solve such problems is to analyze the SIP signaling on VSP side. Without seeing the SIP messages people can only guess.

Troubleshooting problems with Nokia it's necessary to keep in mind a few things:
- Nokia will use STUN automatically, but this may be changed from Nokia VoIP Settings application; depending on NAT device STUN usage may create a problem
- Nokia will always use source port 5060, if another SIP UA behind the same NAT router will use the same port this may create a problem
- Nokia creating INVITE message of quite a big size; this may create a problem with outgoing calls from it through SIP proxy, but this should not be a problem in Voxalot case (as Voxalot is running Asterisk which is B2B UA and not a SIP proxy); disabling codecs and switching to compact headers will not dramatically decrease the size;
- some Nokia models have SIP identity (From: field) hidden by default, this should be changed, http://forum.voxalot.com/voxalot-sup...html#post22139

Last edited by green; 04-07-2009 at 07:08 PM. Reason: accuracy: changed 'will' to 'may'
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Unread 04-07-2009, 09:09 AM   #6
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Quote:
Originally Posted by sleek View Post
It could be that you've mistakenly placed your E71 softphone in DND mode. I often see people do that without knowing so by dialing certain codes or enabling it from their ATAs. In that regard check that your E71 isn't placed on DND in which case incoming calls are routed to your voice mail as you describe.

Second, it could be codec mishandling. It's possible the voice codecs on your E71 to fail to communicate with the server or end clients.

Trying calling different echo tests on different severs, for example:
*266301
*645904
*850301
See if you get any results.

Lastly, because you're behind a router the NAT handling with Eyebeam may not be as good as your E71. Perhaps you should post your settings so we can go from there. As last resort, you'll have to enter the web configuration of your DSL router and do some port forwarding.
It is VERY weird what is happening

First let me assure you it is not related to DND or anything however sometimes from the nokia phone (I've tried from a e71 and e66 same thing) i call the *600 and it works perfectly and sometimes (without changing anything) I dial *600 and i get no sound.

In all cases i cannot call a nokia registered to voxalot behind this dsl router

Quote:
Originally Posted by green View Post
- Nokia will always use source port 5060, if another SIP UA behind the same NAT router will use the same port this will create a problem
I think you might have nailed it i guess this is the error, anyway of changing that ?

Last edited by akhater; 04-07-2009 at 09:21 AM.
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Unread 04-07-2009, 07:12 PM   #7
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Quote:
Originally Posted by akhater View Post
I think you might have nailed it i guess this is the error, anyway of changing that ?
I just changed the wording in my post. This port thing may create a problem and not 100% will. This really depends on the NAT device. Some devices will preserve the port number then others will assign the random port on WAN. Only the 1st case is potentially problematic. Anyway, you can always change a port on another device and let Nokia stay with 5060.
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Unread 04-07-2009, 08:10 PM   #8
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Quote:
Originally Posted by green View Post
I just changed the wording in my post. This port thing may create a problem and not 100% will. This really depends on the NAT device. Some devices will preserve the port number then others will assign the random port on WAN. Only the 1st case is potentially problematic. Anyway, you can always change a port on another device and let Nokia stay with 5060.
and what if i am using 2 nokias in my lan ?
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Unread 04-07-2009, 09:13 PM   #9
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Quote:
Originally Posted by akhater View Post
and what if i am using 2 nokias in my lan ?
Depends on your router & provider.
For example, one of my providers is accepting connections over both TCP & UDP, so I may use 2 Nokia's configured differently.
If your router preserves port then you're in trouble. If not - should be OK. Just connect to any SIP-provider who will show your registration status (address, port, UA) and compare the port with what is configured locally.
Alternatively you may check your eyeBeam log, you will obviously see your local port there and you will also see your external port with rport= statement in the server's reply). We had a long thread on this recently...
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Unread 04-08-2009, 08:11 AM   #10
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I'm not very good in this so I need your help :s

The router I am using is the TG585 v7 and the eyeBeam is working fine, my problem is with the 2 Nokias I have.

What should I check in my router?

If I call *0@proxy01.sipphone.com I get the "you are behind a sip compatible router" message
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