lonal Solutions Australia Pty Ltd
PO Box 220
Walkamin Qld 4872
Phone: (07) 4093 3826
Fax: (07) 4093 3869
Email: andrew.ra...@yuruga.com.au <mailto:andrew.ra...@yuruga.com.au>
Web: www.yuruga.com.au <http://www.yuruga.com.au/>
On 19/04/12 3:42 AM, Stiles Watson wrote:
Well, I'd still l
roduct and I hope it continues to improve and
continues to be used by more companies.
Thank you, Tony, and everyone else, for your help.
Stiles
On 04/19/2012 01:31 PM, Tony Graziano wrote:
On Thu, Apr 19, 2012 at 1:06 PM, Stiles Watson <mailto:wat...@datatek-net.com>> wrote:
t abilities. The thing which is
easy for one, may very very difficult for another.
DNS, DNS, DNS! What beautiful words! On to DNS!
On Wed, Apr 18, 2012 at 5:44 PM, Stiles Watson <mailto:wat...@datatek-net.com>> wrote:
I'm not sure what there is to 'get'. I thought
Picher wrote:
since when would you enter a pin in a manually configured sip phone as
your sip password?
on bria ipad i don't need to enter an outbound proxy or auth name...
but then again, my external dns is setup properly...
mike
On Wed, Apr 18, 2012 at 4:07 PM, Stiles Watson <ma
t;, you need to enter Out. Proxy and
Auth Name
I do not know what the approval process is, but if I can get a
login, I'd be happy to do it. Could also enter settings for
Media5-fone.
Stiles
On 04/18/2012 03:50 PM, Stiles Watson wrote:
Not sure what the de
y and I've not been working in the guts of this system for
months and years so I'm just asking for a little clarity so I can keep
moving this project forward.
Stiles
On 04/18/2012 04:32 PM, Tony Graziano wrote:
Account advanced works unchecked if your DNS is correct.
On Apr 18, 2012 4
#x27;d be happy to do it. Could also enter settings for Media5-fone.
Stiles
On 04/18/2012 03:50 PM, Stiles Watson wrote:
Not sure what the deal was, but I sent the profile AGIAN, set the
password to the SIP password, not PIN and it registered.
Stiles
On 04/18/2012 03:22 PM, Stiles Watson wro
Not sure what the deal was, but I sent the profile AGIAN, set the
password to the SIP password, not PIN and it registered.
Stiles
On 04/18/2012 03:22 PM, Stiles Watson wrote:
I am unable to register via Bria for ipod touch. I'm connected via
wifi on the office network. I get Unauthorized
I am unable to register via Bria for ipod touch. I'm connected via wifi
on the office network. I get Unauthorized (401).
Here are my settings:
Account Name: 295
Enabled: ON
Username: 295
Password: (I've tried both PIN and SIP)
Domain: datatek-net.com
Under "Account Advanced"
Out. Proxy: sipx
Well, I'd still like to get the 3CX iphone client working, but I
downloaded Media5Fone and was able to get it registered and working in a
matter of minutes over both wifi and 3G.
Stiles
On 04/18/2012 12:45 PM, Stiles Watson wrote:
Beat me up all you want, I can take it.
Here are th
do things by DNS... Don't mess with IP's... learn
about SRV records
Mike
On Wed, Apr 18, 2012 at 11:46 AM, Stiles Watson
mailto:wat...@datatek-net.com>> wrote:
Where?
I changed Local PBX IP to sip domain, but it still does not work.
Stiles
On 04/18/201
Where?
I changed Local PBX IP to sip domain, but it still does not work.
Stiles
On 04/18/2012 11:24 AM, Michael Picher wrote:
try sip domain...
On Wed, Apr 18, 2012 at 11:18 AM, Stiles Watson
mailto:wat...@datatek-net.com>> wrote:
What are the settings to get the 3CX softphone w
What are the settings to get the 3CX softphone working with sipX? Tony,
you mention it here and I see that it is listed as a supported softphone
on the wiki:
http://wiki.sipfoundry.org/display/sipXecs/List+of+Features
But no setup is detailed.
The iphone is connected via wifi on the local net
version.
On Tue, Apr 17, 2012 at 11:39 AM, Stiles Watson
mailto:wat...@datatek-net.com>> wrote:
I have a user trying to provision a Bria softphone on a ipod touch
over a remote wifi. I've created a user for him, added the bria
device, assigned the user as a line on the devi
I have a user trying to provision a Bria softphone on a ipod touch over
a remote wifi. I've created a user for him, added the bria device,
assigned the user as a line on the device and sent the profile. I've
instructed him to use the his ext as his user name, his PIN as his
password (via instru
"486 Busy Here"
message and is sent to v-mail.
Stiles
On 04/13/2012 06:09 PM, Stiles Watson wrote:
The config file is not getting rejected because I can see the changes on
the phone after I make them in the sipX interface.
I just did a siptrace of the second call (active call on
to allow MORE THAN ONE call,
> where the limit is placed is on the line itself. I think you need to go back
> and UNDO the setting for 1 on the device and change it back.
>
> On Fri, Apr 13, 2012 at 12:33 PM, Stiles Watson
> wrote:
>> This is exactly what I've done.
>>
ed, set it
there, and click on the mac address -- assign it there. Maybe there
is a better way, but this method does work for me.
*From:*sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] *On Behalf Of *Stiles
Watson
*Sent:* Friday, April 13, 2012 8:00 AM
lto:sipx-users-boun...@list.sipfoundry.org] *On Behalf Of *Stiles
Watson
*Sent:* Thursday, April 12, 2012 8:58 AM
*To:* Discussion list for users of sipXecs software
*Subject:* Re: [sipx-users] Multiple line appearances
That is what I did. I have one line assigned to the phone and the
registration settings as belo
line appearances
That should work fine, it has on every 650 i've ever tried it on.
I've honestly never ever tried it on a 335 however. I generally use
this feature for operator types... and operator types don't use 335's.
Mike
On Thu, Apr 12, 2012 at 12:58 PM, Stiles Wa
t has on every 650 i've ever tried it on.
I've honestly never ever tried it on a 335 however. I generally
use this feature for operator types... and operator types don't
use 335's.
Mike
On Thu, Apr 12, 2012 at 12:58 PM, Stiles Watson
mailto:wat...@dat
t has on every 650 i've ever tried it on.
I've honestly never ever tried it on a 335 however. I generally
use this feature for operator types... and operator types don't
use 335's.
Mike
On Thu, Apr 12, 2012 at 12:58 PM, Stiles Watson
mailto:wat...@dat
firmware 3.2.6.0314 and bootROM 4.2.2.0710
Stiles
On 04/12/2012 01:06 PM, Tony Graziano wrote:
you should also indicate what firmware version your phones are using.
On Thu, Apr 12, 2012 at 12:58 PM, Stiles Watson
mailto:wat...@datatek-net.com>> wrote:
I'm either not commu
per
line key.
On Thu, Apr 12, 2012 at 11:57 AM, Stiles Watson
mailto:wat...@datatek-net.com>> wrote:
That is what I did. I have one line assigned to the phone and the
registration settings as below.
Stiles
On 04/12/2012 11:54 AM, Tony Graziano wrote:
You regi
That is what I did. I have one line assigned to the phone and the
registration settings as below.
Stiles
On 04/12/2012 11:54 AM, Tony Graziano wrote:
You register the SAME LINE on both line appearances and set the limit
to "1" for each.
On Thu, Apr 12, 2012 at 11:51 AM, Sti
According to the sipX book (p 133):
"Most multiple line IP hardware phones allow multiple calls to a single
line. This
can be quite confusing for the average phone user and difficult to deal
with at an
answering position. To remedy this problem it is easier for the user to
have multiple
appeara
get every bit of functionality isn't always
prudent right away EVEN if it is possible. Users need to grasp the
basic differences when migrating from a KEY system. Fortunately you
can put the handsets side-by-side for an acclimation period.
On Wed, Apr 11, 2012 at 12:12 PM, Stiles Watson
ma
2 12:07 PM, Michael Picher wrote:
Those are the only feature codes...
Other features are phone specific.
Did you want a SIP Communications System or a TDM PBX? ;-)
Mike
On Wed, Apr 11, 2012 at 10:40 AM, Stiles Watson
mailto:wat...@datatek-net.com>> wrote:
There is a sample Quick re
Our current system has the ability for any ext to set "Busy Call
Handling." We use this in a very limited case. It is simply the
forwarding of a call when the ext is busy. Does this exist in sipX? It
is not a hunt group per say because if there is no answer it goes
straight to v-mail and is not
Message-
> From: sipx-users-boun...@list.sipfoundry.org
> [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Stiles Watson
> Sent: Tuesday, April 10, 2012 3:55 PM
> To: Discussion list for users of sipXecs software
> Subject: [sipx-users] sipX feature codes
>
>
Is there a central list of all the feature codes in sipXecs? I've
searched "feature codes" in both the wiki and the book, but can not find
a list.
Stiles
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I am saying if you are calling from a voip.ms trunk
to another voip.ms
subscriber it shows the sipx cnam because they DO allow that.
I.e. you call me from voip.ms it shows me your internal
name/line like an internal call.
On Apr 9, 2012 2:37 PM, "Sti
On Mon, Apr 9, 2012 at 12:02 PM,
Stiles Watson <wat...@datatek-net.com>
wrote:
OK, I understand. I
see the name in the voip.ms CDR.
I'll contact voip.ms t
:sipx-users-boun...@list.sipfoundry.org]
On Behalf Of Stiles Watson
Sent: Monday, April 09, 2012 11:48 AM
To: sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] Caller Id showing wrong
display
Then why have the Display name field?
On 04/09/2012 11:40 AM, Michael Picher wrote:
names must come from provider...
otherwise i could change my text to read whatever i want...
On Mon, Apr 9, 2012 at 11:36 AM, Stiles Watson <mailto:wat...@datatek-net.com>> wrote:
I'm
I'm using voip.ms for my ITSP. In the gateway settings I have "Specifiy
Caller ID" checked and the 1+10 digit number in the "Caller ID" field
and 'Datatek Ink' in the "Display name" field. The correct number is
displaying, but the display name for the caller id is coming across as
'Starnet Inc
Thanks
On 04/06/2012 02:00 PM, Tony Graziano wrote:
If you searched the sipfoundry wiki...
EFK would have returned this.
http://wiki.sipfoundry.org/display/sipXecs/Polycom+Phone+Customization
On Fri, Apr 6, 2012 at 1:57 PM, Stiles Watson <mailto:wat...@datatek-net.com>> wrote:
the right
direction.
Stiles
On 04/06/2012 01:46 PM, Tony Graziano wrote:
UNLESS you write the EFK to enable it, the web portal is required.
On Fri, Apr 6, 2012 at 1:40 PM, Stiles Watson <mailto:wat...@datatek-net.com>> wrote:
We are a small office with some remote employees. On
550's & 650's that lets users set the system call
forwarding as well as a hotelling application.
Mike
On Fri, Apr 6, 2012 at 12:58 PM, Stiles Watson
mailto:wat...@datatek-net.com>> wrote:
First, after searching both the sipx book and the wiki, it
First, after searching both the sipx book and the wiki, it looks like
call forwarding is only available via the web portal, is this correct?
My users are used to doing it from their phones, is this possible from a
Polycom 335?
Second, is there a way to remove the superadmin user from the direct
When using either SPA 2102 or 3102 for remote phones, do they have to be
used behind a VPN? Since tftp is not available in this scenario, can they
use ftp to download a new profile?
I read
http://wiki.sipfoundry.org/display/sipXecs/Linksys+SPA-series+ATA+Devices,
but it does not mention anythin
protocol. this is why ftp is also a choice. you
m,ust nat ftp through your firewall.
On Thu, Apr 5, 2012 at 12:30 PM,
Stiles Watson wrote:
I'm trying to get a Polycom 335 to work remotely
without VPN. The phone registers and everything works except contacting the
boot server.
Registr
I'm trying to get a Polycom 335 to work remotely without VPN. The phone
registers and everything works except contacting the boot server.
Registration Example:
sip:1...@mycompany.com
On the phone, under the
ServerMenu,
* ServerType: TrivialFTP
* ServerAddr:
tftp%3a%2f%2f
If so, are you doing it over VPN?
Stiles
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k well and we know
they work...
Mike
On Wed, Apr 4, 2012 at 12:01 PM, Stiles Watson
mailto:wat...@datatek-net.com>>
wrote:
Is there a better SPA type device that anyone recommends?
Stiles
Wed,
Apr 4, 2012 at 11:05 AM, Stiles Watson wrote:
Either one of these may be
a good option for my specific case. Do either the SPA 2102 or 3102 work
well remotely over VPN? What about without VPN?
Stiles
On Wed, 4
Apr 2012 06:51:33 -0400, Michael Picher wrote:
You can also run Bria on
Wifi, or hook up an analog gateway and use a standard analog/dect wireless
phone. Mike
On Tue, Apr 3, 2012 at 5:47 PM, Stiles Watson wrote:
Anyone recommend the Snom M9 for a remote ext? The remote worker only
needs to make and receive calls and have access to v-mail. A DID would
be
assigned
Anyone recommend the Snom M9 for a remote ext? The remote worker only
needs to make and receive calls and have access to v-mail. A DID would
be assigned directly to this phone.
I see there is a template for the eol m3. Is the m9 setup similar?
Stiles
dged sword with
XMPP. XMPP with openfire can only be the sipdomain (not an alias).
Since most of my customers ahve XMPP service hosted elsewhere, it's
more desirable to use a subdomain as the sipdomain. We can always
alias the sipdomain (just not the xmpp domain).
On Mon, Apr 2, 2012
use a subdomain as the sipdomain. We can always
alias the sipdomain (just not the xmpp domain).
On Mon, Apr 2, 2012 at 5:04 PM, Stiles Watson <mailto:wat...@datatek-net.com>> wrote:
So as an example:
I want to use the company domain for the sip domain and I want the
sip sub net
mpany.com <http://sipx.mycompany.com>. If you
want to add another server at a later time or use XMPP to
mycompany.com <http://mycompany.com> in the future, I would use
mycompany.com <http://mycompany.com> (which should be the default
response).
Mike
On Mon, Apr 2, 2012 at 5
threads, they all point back to the same
issue. DNS.
This is how I would do i if I were you, but I am not you (since people
tire of my "if it were me...").
I do this the same each and every time. I never have registration
issues. I never have MWI issues and I never have i
d then try to move to a host name, that's different...
Mike
On Fri, Mar 30, 2012 at 3:04 PM, Stiles Watson <mailto:wat...@datatek-net.com>> wrote:
I'll be happy to fix whatever is broken, but please help me
understand first.
Are you saying that the pro
m do I have to do another reinstall
or is there another way to switch over?
Stiles
On 03/30/2012 02:21 PM, Tony Graziano wrote:
Its the fact you are registering to the host name instead of the domain.
Fix your DNS.
On Mar 30, 2012 1:41 PM, "Stiles Watson" <mailto:wat...@datatek-
I just noticed that since I upgraded sipX to 4.4 and my Polycom IP 335
phones to firmware 3.2.6 and BootROM 4.2.2, I no longer get a v-mail
notification on the Phones. The message indicator LED does not flash and
there is no indication on the display except "1 new missed call".
I do not know en
The phone registrations are as follows:
sip:1...@sipx.datatek-net.com
sip:1...@sipx.datatek-net.com
On 03/28/2012 05:38 PM, Stiles Watson wrote:
Numeric only. I'm trying to place a call from ext 145 to ext 141. They
are on the same subnet and they both got their IPs from the DCHP
s
Numeric only. I'm trying to place a call from ext 145 to ext 141. They
are on the same subnet and they both got their IPs from the DCHP server
running on the sipX server.
Stiles
On 03/28/2012 05:34 PM, Tony Graziano wrote:
I don't know why anyone would do subscriber lines that weren't numeri
in wayback machine multiple times for you.
You may also consider updating ONE phone to 3.2.6 and bootrom 4.3.1
and test just it.
On Tue, Mar 27, 2012 at 4:20 PM, Stiles Watson <mailto:wat...@datatek-net.com>> wrote:
Todd,
No, I've not changed any port info
ware
*Subject:* Re: [sipx-users] voip.ms config
That looks wrong. The destination port for an outbound call should be
5060 not 5080. If you used the stock template the firewall is he issue
here.
On Mar 27, 2012 3:38 PM, "Stiles Watson" <mailto:wat...@datatek-net.com>> wrote:
I
/27/2012 03:49 PM, Tony Graziano wrote:
That looks wrong. The destination port for an outbound call should be
5060 not 5080. If you used the stock template the firewall is he issue
here.
On Mar 27, 2012 3:38 PM, "Stiles Watson" <mailto:wat...@datatek-net.com>> wrote:
. #1 above only appears when there is an active call.
Stiles
On 03/27/2012 12:25 PM, Gerald Drouillard wrote:
On 3/27/2012 12:03 PM, Stiles Watson wrote:
This is where one swallows one's pride The way I was entering
data caused the drop-down to not be displayed.
To keep this short
behavior is different in 4.4. Updating would
keep us all from having to use a way back machine.
On Mar 27, 2012 12:05 PM, "Stiles Watson" <mailto:wat...@datatek-net.com>> wrote:
This is where one swallows one's pride The way I was entering
data caused the drop-
what options you have there.
On Mon, Mar 26, 2012 at 6:10 PM, Stiles Watson
mailto:wat...@datatek-net.com>> wrote:
It is not there. I've tried Devices>Gateways>Add new
gateway... a dozen times. I've restarted all the services,
I've re
configuration screen. 4^th item down
is the templates selection box...
*From:*sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] *On Behalf Of *Stiles
Watson
*Sent:* Monday, March 26, 2012 2:05 PM
*To:* Discussion list for users of sipXecs s
Walking through Tony's voip.ms how-to. All my notes are delimited by
---> <---and are in /_italics and underlined_/.
**
*Dealing with Step 3, online with voip.ms*
At the voip.ms portal:
Main Menu > Account Settings (for a main account, not subaccounts)
>Account Restrictions
Adjust the call ti
Well, I looked at the voip.ms registration status and it lists my WAN IP
and port 5080 so it does not seem like a port issue.
On 03/26/2012 12:30 PM, Gerald Drouillard wrote:
On 3/26/2012 11:00 AM, Stiles Watson wrote:
Anyone else have an idea why I lose audio when retrieving a call from
hold
Hmm, That's likely. Here's some info from voip.ms:
SIP: 5060 UDPRTP Range: 10001-2 UDP
Is there a way to make voip.ms match sipx or do I have to change sipx to
match voip.ms (and change my finewall)?
When I set up voip.ms, I used this how-to from Tony. However, some
things have chang
Anyone else have an idea why I lose audio when retrieving a call from
hold or canceling a transfer?
Stiles
On 03/21/2012 05:37 PM, Stiles Watson wrote:
Yes, under Account Settings-->Inbound Settings, I set Device Type to
"IP PBX Server, Asterisk or Softswitch". Is this the only
voip.ms <http://voip.ms> did you change the account to be
'Asterisk or SIP PBX' (i'm paraphrasing).
mike
On Wed, Mar 21, 2012 at 5:04 PM, Stiles Watson <mailto:wat...@datatek-net.com>> wrote:
We'll I signed up for voip.ms <http://voip.ms> and set up th
fer
which does not work.
Also, many of the calls through voip.ms are 'active' according to the
CDR. I'm guessing these are the lost trnasfers and holds? Some of them
have been active for over 30 min after I end the call.
Stiles
On 03/21/2012 12:29 PM, Stiles Watson wrote:
While s
k are incoming calls
answered by the AA can not transfer to an ext. The CDR says the call was
transfered, but it is. Calling out works, and ext to ext works.
Stiles
On 03/21/2012 11:18 AM, Stiles Watson wrote:
OK, thanks. DNS is running on the sipXecs server.
I re-installed and used the fully qua
voip.datatek-net.com:SipRouter-11:B6D77B90:SipXProxy:"SipUserAgent::send
outgoing call 1"
"2012-03-21T15:01:41.158371Z":770:KERNEL:ERR:voip.datatek-net.com:SipClientTcp-377::SipXProxy:"OsMsgQShared::doSendCore
message send failed for queue 'SipTcpServer-3'
stall and do it with A-Records
Be aware if you use A-Records you can not just add servers to the
cluster. This assumes SRV records.
Mike
On Mon, Mar 19, 2012 at 2:31 PM, Stiles Watson <mailto:wat...@datatek-net.com>> wrote:
This is a follow up question to a problem I sub
nd do it with A-Records
Be aware if you use A-Records you can not just add servers to the
cluster. This assumes SRV records.
Mike
On Mon, Mar 19, 2012 at 2:31 PM, Stiles Watson <mailto:wat...@datatek-net.com>> wrote:
This is a follow up question to a problem I submitted abou
This is a follow up question to a problem I submitted about 2 months
ago. Long story short, I reinstalled sipxecs 4.2 and made the sipxecs
server the DHCP and DNS servers as well. This solved some of my issues,
but not all. I still am unable to transfer incoming calls to any
extension. I can di
As tony said, enable some rate limiting on your firewall or don't open
those ports if you don't need them.
On Jan 16, 2012 4:29 PM, "Stiles Watson" <mailto:wat...@datatek-net.com>> wrote:
Long story short, our sipx project was pushed to the back burner
abou
44::SipXProxy:"OsMsgQShared::doSendCore
message queue 'SipTcpServer-3' is over half full - count = 91, max = 100"
Any clues?
Stiles Watson
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> using 57600 which is about 16 hours. Should be enough :-)
>
> On Mon 13.Dec.10 15:24, Stiles Watson wrote:
>
>> How are you working around it?
>>
>> I like Flowroute too. I use them for my home ITSP with a Cisco SPA2102
>> with no timeout problems.
>>
to another ISTP. I like Flowroute,
> but perhaps they are not the best match for Sipx.
>
> Dan
>
> On Thu 09.Dec.10 13:21, Stiles Watson wrote:
>
>> Dan,
>>
>> Any solution on this?
>>
>> Stiles
>>
>> Dan McDaniel wrote:
>>
Dan,
Any solution on this?
Stiles
Dan McDaniel wrote:
> Yes, we are still working on this, but it's not resolved yet. Turns out
> that 30 minutes (1800 seconds) is the time in the Session Timer Interval
> box on the gateway setup screen. It controls what goes into the Session
> Expires header in
Which wireless SIP phones which work with sipX and what are your
experiences with them?
We do not employ wireless in our offices so I'm mostly interested in
non-wifi options like DECT or even wireless headsets connected to the
desktop phone, but a thread with experiences and options of all wire
Dan,
Did you ever hear back from flowroute on this issue? I also have calls
dropping at 30 min. I've created a support ticket for this as well.
Stiles
Dan McDaniel wrote:
> On Mon 06.Sep.10 09:04, dan wrote:
>
>> I've been having a problem with calls dropping. The ISTP (Flowroute)
>> sends a
l.com/products.html#Softphone
>
> If it works, report back to the list the results.
>
> -Original Message-
> From: sipx-users-boun...@list.sipfoundry.org
> [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Stiles Watson
> Sent: Wednesday, October 06, 2010
> Is it OK when he is using exactly the same setup in the states?
> If not then probably the echo is caused by his setup.
> Also, is the sound OK when he is calling a sip user on the same SipX?
>
> What is he using as Audio device?
> Headset should be OK, Clearone Chat50 should be OK
M Broome
> *CEO*
> SATEL, inc
> (800) 591-7033
>
>
>
>
>
>
> On Tue, Oct 5, 2010 at 10:36 AM, Stiles Watson <mailto:wat...@datatek-net.com>> wrote:
>
> I've got a VIP overseas using X-lite to place calls over the public
> Internet
I've got a VIP overseas using X-lite to place calls over the public
Internet back to the States via our sipX. Calls go through fine, but he
says that the some callees tell him that there is a bad echo (all hear
themselves to some degree). All outbound calls go though our ITSP
(flowroute). All c
:24 2010
> Subject: Re: [sipx-users] SRV records for ftp
>
> I've talked to the ISP and they say that they do not block any ports.
>
> Stiles
>
> Stiles Watson wrote:
>
>> Yes. In this case the service is not used.
>>
>> Stiles
>>
>> Tony G
I've talked to the ISP and they say that they do not block any ports.
Stiles
Stiles Watson wrote:
> Yes. In this case the service is not used.
>
> Stiles
>
> Tony Graziano wrote:
>
>> Does the cable modem provider offer a voice service?
>> ==
->NAT is set to the local WAN
> * Under Internet Calling-->NAT both boxes are checked
> * Intranet Domains is set to the correct domain
> * Intranet Subnets is set the company's LAN subnet in the following
> format: xxx.xxx.xxx.0/24
> * SIP ALG is turned off on the local son
SIP ALG is turned off on the local sonicwall
* firewall rules and NAT rules are in place
Any clues?
Stiles
Tony Graziano wrote:
>
>
> On Mon, Sep 20, 2010 at 2:19 PM, Stiles Watson <mailto:wat...@datatek-net.com>> wrote:
>
> See response below:
>
> Tony Graz
ked
* remote users are enabled
* For intranet subnets, does the remote user's subnet have to be
added? I'm assuming so, but I just want to be sure.
Still does not work, but I'll keep digging.
Stiles
>
> On Mon, Sep 20, 2010 at 1:37 PM, Sti
he registration request is getting to sipx. I turned the logging level
>> to DEBUG, restarted the services and executed following:
>>
>> tail -f /var/log/sipxpbx/sipXproxy.log | grep "REGISTER sip" | grep
>> "1...@datatek-net.com" > regdebug141.log
>>
executed
>
> grep -i received regdebug141.log | wc -l
>
> with a result of '9'. When I tail /var/log/sipxpbx/sipregistrar.log I
> see the following:
>
> "2010-09-20T15:19:11.048702Z":33020:AUTH:DEBUG:sipx.datatek-net.com:SipRegistrarServer:B6C81B90:SipRegist
T15:19:11.048702Z":33020:AUTH:DEBUG:sipx.datatek-net.com:SipRegistrarServer:B6C81B90:SipRegistrar:"SipRegistrarServer::isAuthorized
fromNameAddr='\"Stiles
Watson\";tag=1F8AD21E-973871A1',
toUri='sip:1...@datatek-net.com', realm='datatek-net.com'"
"2010-09
to do.
> Can you get another IP and add it to the firewall (even if just for
> ftp)...?
>
> On Fri, Sep 17, 2010 at 6:26 PM, Stiles Watson wrote:
> Thanks, you are a wealth of info! I'll try the several options you've
> given me.
>
> FYI, I had an Aastra 673
based routers, they're a pain.
>
> http://www.rfc-editor.org/rfc/rfc3489.txt
>
> It makes me need a drink, and its why I use FTP for remote phones.
>
> There is a way to get that to work, but you must have the required
> items (port translation, and that pattern is full).
&g
y to try?
So this means you can test with what you got but rearrange the
firewall, push your configs, and then change it back... or get another
public IP on your firewall for this...
On Fri, Sep 17, 2010 at 5:19 PM, Stiles Watson <mailto:wat...@datatek-net.com>> wrote:
OK
OK Tony, shoot me down:
I'm attempting to do what you suggested and use FTP instead of TFTP for
remote provisioning the Polycom IP 335. The problem is that we already
use FTP and we can not move our customer facing FTP to another port. I
figured I could just configure the phone to use ftp on an
We already give ftp access to our clients and its directed to a
different server. I'm assuming I can pick a different port and just PAT
the traffic to port 21 on the sipX server.
Stiles
Tony Graziano wrote:
> Don't do that. Wipe the phone config. NAT port 21 at the sonicwall to sipx.
>
> Go to
I'm able to get a couple of different softphones to remote register with
sipX. However, I'm unable to get my Polycom IP 335 to register remotely.
Below are my settings.
Remote Firewall/DHCP config:
* DD-WRT OS runing on Linksys WRT54GL
* SPI Firewal disabled (for this test)
* DHCP Server enabled
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