Hello Mark:
" p.s. monitored loop time is ~140 to 1100ms... is that a lot? it sounds
like
about a second to me....I can't actually try client's phone till tues...
these run thru two vpn d-links. should I be worried, ya think? "
Considering my staff in Philippines has an average of 350-450ms at the
worst... right here in Ontario if you fluctuate between 140-1100ms, I would
say is pretty damn bad.
No sip client on our servers exceed 100ms delay. Usually MOST fluctuate
between 30 & 90ms.... on the average, its about 50ms, on my side. If for
any reason someone exceeds the 100ms threshold, then we know the client's
internet connection is not the best.
We recommend our clients have dedicated internet connection for their VOIP
needs, free from office computer network be it LAN/VPN etc. Just my 2 cents
:).
Cheers!
Reza.
----- Original Message -----
From: "Mark Borg" <[EMAIL PROTECTED]>
To: <[email protected]>
Sent: Sunday, May 20, 2007 10:39 PM
Subject: Re: [on-asterisk] cannot register SIP offsite thru vpn....
On Sun May 20 2007 22:29:51 Reza - Asterisk Enthusiast wrote:
do a "sip show peers" and it will show you the lag time of your off site
phone.
Cheers!
Reza.
----- Original Message -----
From: "Mark Borg" <[EMAIL PROTECTED]>
To: <[email protected]>
Sent: Sunday, May 20, 2007 10:13 PM
Subject: Re: [on-asterisk] cannot register SIP offsite thru vpn....
> On Sun May 20 2007 21:45:43 Reza - Asterisk Enthusiast wrote:
>> Mark:
>>
>> There is likely a problem in your configuration file. The solution
>> is
>> quite simple, but the information you have provided is not enough to
>> solve
>> your issue. The debug messages you provided merely shows the
>> registration has failed. We need to look at 2 things:
>>
>> 1. Your SIP.CONF file ... not the entire file, but the segment that
>> has your 6010 settings. I suspect you have it under your [6010]
>> context.
>>
>> 2. Your Aastra configuration and model number and the exact location
>> where
>> you entered your credentials. That is "Global SIP" and/or settings
>> under "Line 1" of your phone. We need to see the exact entries
>> except
>> the password.
>>
>> Its clear from the message that your Aastra is connecting to your
>> Asterisk
>> server and for some reasons it is not accepting your registration.
>> Additionally please check the firmware version of your phone.
>>
>> Last Friday we were at a clients who purchased close to a dozen phones
>> (brand new) but had firmware dated May 2005. We upgraded it to
>> firmware date 2007 version. 1.4.something... and it resolved their
>> issues without
>> us having to touch any configurations on their phones!
>>
>> Waiting for your settings to be able to properly analyze.
>>
>> Cheers!
>> Reza.
>>
>>
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>
> Hi Reza, thanks for the response!
> After 3 or 4 hours looking at the same lines (sleepy?) I set the domain
> to '192.168.0.,"mycontext"... 192.168.2.,"mycontext"'; now domains are
> OK. I
> suppose I "knew" about not having last octet for a domain, just didn't
> see
> it.
> Anyways, everyone is now registered and reachable, but....
> Placing a call to the off-site extension (6006) shows 'busy'.
> it is via vpn... maybe too much lag or ?
> this is the debug of the call, if you wanna see it...
> -- Executing [EMAIL PROTECTED]:1] Dial("Zap/1-1", "SIP/6006|1") in
> new stack
> -- Called 6006
> -- SIP/6006-0820cfd8 is ringing
> -- Got SIP response 486 "Busy Here" back from 192.168.0.226
> -- SIP/6006-0820cfd8 is busy
> == Everyone is busy/congested at this time (1:1/0/0)
> thanks for the time!!!
> markborg
>
>
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thanks again Reza....
didn't notice that "1" in the dial timeout.... jeez i need more coffee.
fixed up extensions.conf, all looks pretty good for tues a.m.
p.s. monitored loop time is ~140 to 1100ms... is that a lot? it sounds
like
about a second to me....I can't actually try client's phone till tues...
these run thru two vpn d-links. should I be worried, ya think?
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