Gary Luca garyluc...@gmail.com wrote on 15-07-2010 05:02:40:
Ok. So I just got a whole TON of replies with questions to answer. Here
goes:
Nathaniel...
DNS/DHCP at the remote site are handled by the Westell ADSL2 modem.
It deals out just IP, Subnet mask, gateway (itself) and DNS (also
You might find your internal firewall may have an issue as well passing
audio if it cannot do symmetrical NAT (which I doubt).
So once you get DNS setup where the phones will register you should try
making a call and if you get no (or one-way audio), you should stop and
address the firewall
I may have spoke too soon...
I see you can load dd-wrt on the actiontek...
http://www.dd-wrt.com/wiki/index.php/Supported_Devices#Actiontec
http://www.dd-wrt.com/wiki/index.php/Supported_Devices#ActiontecQuestion
is what revision and is it in the supported devices list or on the HCL list,
only
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What you are trying to do is unsupported
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You'll want to look in
Avaya is fully committed to supporting sipXecs as an open source
project, and will continue to encourage contributions to this project
from the community as well as continue with its own contributions.
However, communications regarding the direction of sipXecs, feedback
from the community on
Hi Al,
First, I started with 3.6 and have had a lot of fun as well.
Keep up the good work (development)!
Could you be a little bit more clear about the future?
If I understand it correctly SipXecs is moving from sipfoundry to openscs.
That leaves sipfoundry with ??, or will they develop their
Paul,
SIPfoundry isn't going anywhere and I expect Ezuce will become the
primary supporter along with anyone in the community. I am sure we will
see some communication on that today.
The code was all contributed to SIPfoundry under LGPL so it can be used
without any issue.
Al
Hi Al,
As Al (Al Campbell at Avaya) has mentioned on the sipx-dev list about the
impending change to openscs for avaya, I would like to point out I currently
have had very favorable results thus far with the alternative community
build hosted by eZuce.
I think things will become clear in a week or two
Avaya has expressed a desire to control the messaging of their project and
to control the roadmap, hence the fork from avaya to build upon sipxecs via
the openscs community they are unveiling.
Does the plan include the open source version having less features than the
commercial version?
On Thu, Jul 15, 2010 at 9:30 AM, m...@grounded.net m...@grounded.netwrote:
Avaya has expressed a desire to control the messaging of their project
and
to control the roadmap, hence the fork from avaya to build upon sipxecs
via
the openscs community they are unveiling.
Does the plan
Well, technically Avaya is forking the code. There will now be an Avaya
openscs codebase and a SIPFoundry sipx codebase.
Users will need to choose between reporting bugs, viewing wiki's or roadmaps to
either follow and install the Avaya project or the SIPFoundry project. Or
report to both
I think I detailed as much as can be detailed for now.
Please see:
http://forum.sipfoundry.org/index.php?t=msgth=13784start=0S=622c9c20470ef587cfa7003c839da7ef
On Thu, Jul 15, 2010 at 8:52 AM, Paul Herron p...@sagecraftsmen.com wrote:
So, from a practical, sipx-users standpoint, it sounds
Tony Graziano tgrazi...@myitdepartment.net wrote on 15-07-2010 15:41:09:
On Thu, Jul 15, 2010 at 9:30 AM, m...@grounded.net m...@grounded.net
wrote:
Avaya has expressed a desire to control the messaging of their project
and
to control the roadmap, hence the fork from avaya to build upon
I think mostly that is right.
I have asked for and think we will get a more concise statement from eZuce
(probably sometime today) to help clarify their part and perhaps the name
thing.
What's in a name? That which we call a rose
By any other name would smell as sweet.
I am not so clear on the
Paul Scheepens pscheep...@epo.org 07/15/10 10:15 AM
Both versions will be GPL, so both can borrowcode from each other.
Sipxecs is currently LGPL. Wether or not Avaya keeps openscs under a
compatibile opensource license is up to them. I believe SIPFoundry is not
changing their license for
What's in a name? That which we call a rose
By any other name would smell as sweet.
I vote for SipX (is sipxecs already formally sipx?), since, everyone already
calls it that :).
I'm just an end user, with intentions/hopes of being able to contribute to
features in the future, but for now,
A long time ago, in a land far away, it was the best of times, it was
the worst of times.
*BSD, Linux, opensource, the ability to port software and freely run
mail servers, web servers, database servers helped allow the internet to
grow.
Most of us were working day jobs to pay for our
On Thu, Jul 15, 2010 at 10:56 AM, m...@grounded.net m...@grounded.netwrote:
What's in a name? That which we call a rose
By any other name would smell as sweet.
I vote for SipX (is sipxecs already formally sipx?), since, everyone
already calls it that :).
Let's see if we have to deal with
Hello
I would like to know how can identifiy between a call that teminated
prematury by a technical issue an a call that was terminated by de
caller (hangup).
Thanks in advance
Miguel Díaz
___
sipx-users mailing list sipx-users@list.sipfoundry.org
A call trace would show the BYE and where it came from.
Tony Graziano, Manager
Telephone: 434.984.8430
Fax: 434.984.8431
Email: tgrazi...@myitdepartment.net
LAN/Telephony/Security and Control Systems Helpdesk:
Telephone: 434.984.8426
Fax: 434.984.8427
Helpdesk
First, let me express a big thank you to Al and the Nortel team. Under the
circumstances the dialogue and willingness to do the right thing over the
past months and weeks has been great. There is no fight going on just to
make sure everyone understands that. We are glad that we finally got to
If I have a caller form a PSTN TDM calling to a call center with VoIP
SIP phones answering those calls, is ti posible to ditinguish between
TDM Calls hangedup and TDM lost call due PSTN or VoIP technologies.
In the SIP Trace will appear the BYE issued by the PSTN TDM Caller?
Thanks in advance
Then you should look at your logs on your pstn gateway.
2010/7/15 Miguel Jesús Díaz Macedo miguel.diazmac...@gmail.com
If I have a caller form a PSTN TDM calling to a call center with VoIP
SIP phones answering those calls, is ti posible to ditinguish between
TDM Calls hangedup and TDM lost
I just wanted to let folks know this is not about eZuce taking Avaya's
place. That wouldn't be very useful. With the copyright assignment
agreement out of the way, everyone one is on the same playing field.
eZuce contributes because we found a way to profit in a community
driven, open source
Tony,
Thanks for the XX tracker..
Just to Clarify, I was dealing strictly at the .zip file level using the
sipXconfig. At the linux level just (ls -al) etc. on the .../tfptroot to
see what was there, and a (cat xx) to view ReadMe.txt).
I wanted to have in Device Files one Polycom entry
I would suggest deleting the entry for the one that gives you a problem via
sipxconfig first. Then I'd suggest making sure those zip files are gone from
the tftp directory. Then I'd try to upload the files again and check their
ownership/permissions and try to activate it again.
Ok. So. I'm trying to remotely re-provision the phone and downgrade the
firmware. I figured i'd try TFTP first since I know that works in the local
office. So I forwarded port 69 on the local router to the internal IP of my
sipX. Went into the network setup menu on the phone and defined
Forward port 21 and use ftp instead. I don't necessarily think this design
will work at all for you though unless you change out your firewall type at
the sipxecs side. You should get it to provision and register, but I'm sure
you'll have audio issues with that fios box.
Ok. So. I'm trying to remotely re-provision the phone and
downgrade the firmware. I figured i'd try TFTP first since I
know that works in the local office. So I forwarded port 69
on the local router to the internal IP of my sipX. Went into
the network setup menu on the phone and
I just applied updated from 4.0.1 to 4.2.0 on a copy of my production
server. Everything appears to be working as expected. However, I did get
some warnings and errors and wanted to make sure everything is OK.
Before applying the update, I updated my sipxecs.repo with the following:
Its not an issue. There is a JIRA I opened on that. If it were me, nd its
not, I would have bumped to 4.0.4 before going to 4.2.
Tony Graziano, Manager
Telephone: 434.984.8430
Fax: 434.984.8431
Email: tgrazi...@myitdepartment.net
LAN/Telephony/Security and Control
Ok. We are making progress. It APPEARS as though the phone is now
communicating with the sipX via FTP. I deleted and recreated one of the
remote phones in sipX, changing the line configuration from what it used to
be (so I could actually SEE a change from the old configuration). The
Deactivate the old one. Activate the new one. IF the phone is registered
send it its profile and tell it to reboot (or reboot the phone manually).
The phone should auto detect the change and load the firmware accordingly.
There's no sipx magic here, just phone bootrom logic as long as its
We are using 5 digit user extensions.
I tried changing caller ID fields on the User, and on the gateway to no avail.
I know I'm missing something in front of my nose, but can't get sipX to send
out 10 digits for the sipX calling extension caller ID.
Can someone tell me what I'm missing?
OK.
The firmware, bootrom, etc have nothing to do with any specific configs related
to a phone, phone group, line, or user. A phone should pull the firmware simply
from getting the correct options from dhcp. You don't even need the phone setup
in sipx or need to have a user assigned to it.
On 7/15/10 5:02 PM, Smith, Laura M wrote:
We are using 5 digit user extensions.
I tried changing caller ID fields on the User, and on the gateway to no avail.
I know I'm missing something in front of my nose, but can't get sipX to send
out 10 digits for the sipX calling extension caller ID.
Well that's what i FIGURED. But I set the boot server properly in the phone
(I did so manually, as the remote end DHCP is very simple and doesn't
provide option 66), which is evidenced in the fact that the phone is pulling
the config properly. But the firmware isn't downloading.
I even took the
Thanks for the quick response!!
I was pretty sure I had tried that exact config, but for good measure I tried
it again.
Same result. Only 5 digits are sent.
The extension is 12099. In the gateway, I selected Transform Extension,
added caller ID prefix 40425, and tried first with 0 (keep all
Can you put these phones on a vlan that relays dhcp requests to your sipx
server? Not exactly what you are looking for, but that is how I handle it.
-Original Message-
From: Gary Luca garyluc...@gmail.com
Sender: sipx-users-boun...@list.sipfoundry.org
Date: Thu, 15 Jul 2010 17:11:49
To:
There are no managed switches anywhere on the network, so VLANs aren't
really possible. Also, the sipX doesn't handle DHCP for the network. It is
handled by the Server 2008 box.
On Thu, Jul 15, 2010 at 5:33 PM, Matthew Kitchin (Public)
mkitchin.pub...@gmail.com wrote:
Can you put these
Sorry if I'm beating a dead horse. I'm on a bberry and can't see the history on
this thread. Have you run the pre flight test app on a pc on the same network?
-Original Message-
From: Gary Luca garyluc...@gmail.com
Sender: sipx-users-boun...@list.sipfoundry.org
Date: Thu, 15 Jul 2010
Got it. I removed all of the device file profiles from the sipX and then
re-added the 3.1.3c one. So now provisioning works 100%.
Problem 1 solved.
Now back to the real problem of it not registering. I'm going to have the
company hosting our external DNS add in the necessary records. Though
you failed to explain what type of gateway you are using. That has
EVERYTHING to do with it. An analog gateway CANNOT do it, a PRI can only if
the carrier has it allowed. siptrunks depend on the provider.
For grins and giggles remove the callerid from group and gateway and set a
10 digit number
Hello alid
How can I terminate SIP registrations manually?, I have restarted de
registar service, but I still see the registrations as not expired
Thanks in advance
Miguel Díaz
___
sipx-users mailing list sipx-users@list.sipfoundry.org
List Archive:
Don,
The other times I have problems with this, there have been files or
directories that may have been left behind. If there is any sort of problem
un-zipping the firmware or bootrom you will also get this sort of error.
What I typically do to resolve this is disable all archives and then go
I'd be curious to see if xlite actually WORKS besides registering. I am
pretty sure your actiontek will not handle NAT properly at the sipx side, so
beating your dead horse only makes you tired and sore.
You may not ever get it working with the correct firewall because your
public DNS
I DID IT
It is the STUPIDEST little thing.
So in the phone group setup, I was specifying the outgoing proxy in the SIP
Servers screen. Just for the hell of it, I decided to try ALSO specifying
this information under the Lines - Registration screen. And VOILA! It
works.
I definitely feel
I found that even though I was using bootROM 4.2.2 in both entries, I
had to fully delete the bootROM from the upload page. Essentially
leaving 3.1.3revC Device Files entry as an empty shell (no zip files,
references for SIP and bootROM at browse). Then I could activate the
other (i.e. 3.2.3) for
Maybe try opening an FTP session and try to download those Config files?
Or, do a PCAP to see what is failing at the server or the phone. You may
need to do a stare and compare between the two of them.
From: sipx-users-boun...@list.sipfoundry.org
Er. If you say so. I am somewhat doubtful that audio would work in all
cases. Registering is only a babystep.
On 7/15/10, Gary Luca garyluc...@gmail.com wrote:
I DID IT
It is the STUPIDEST little thing.
So in the phone group setup, I was specifying the outgoing proxy in the SIP
Servers
Oh Tony. I tested it and everything. Made an internal call out. External
call out. Internal call in. EVERYTHING WORKS! I'm beyond ecstatic! lol
On Thu, Jul 15, 2010 at 6:56 PM, Tony Graziano tgrazi...@myitdepartment.net
wrote:
Er. If you say so. I am somewhat doubtful that audio would
Must be a big baby...
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Gary Luca
Sent: Thursday, July 15, 2010 7:04 PM
To: sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] Polycom IP 430 won't register remotely
Oh Tony. I tested
If static nat isn't supported at the other end it will systematically
break...
Tony Graziano, Manager
Telephone: 434.984.8430
Fax: 434.984.8431
Email: tgrazi...@myitdepartment.net
LAN/Telephony/Security and Control Systems Helpdesk:
Telephone: 434.984.8426
Fax:
Yes, indeed, Nate. That baby took a HUGE leap. haha
On Thu, Jul 15, 2010 at 7:05 PM, Nathaniel Watkins
nwatk...@garrettcounty.org wrote:
Must be a big baby…
*From:* sipx-users-boun...@list.sipfoundry.org [mailto:
sipx-users-boun...@list.sipfoundry.org] *On Behalf Of *Gary Luca
*Sent:*
Tony Tony Tony. Such the pessimist! ;-)
I'll keep an eye on it over the next few days and see what happens. Either
way I'll post back and let you know how it's holding up.
-G
On Thu, Jul 15, 2010 at 7:06 PM, Tony Graziano tgrazi...@myitdepartment.net
wrote:
If static nat isn't supported
Past experience tells me it would pose a problem with audio. The moment a
port is being used by another app all bets are off...
I'm just saying this so I don't have to do the I told you so later.
Tony Graziano, Manager
Telephone: 434.984.8430
Fax: 434.984.8431
Email:
Don't let him fool you - you'll get one of those later as well :)
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Tony Graziano
Sent: Thursday, July 15, 2010 7:18 PM
To: garyluc...@gmail.com;
Nope. I promise. Instead I will ask who told them that will_work to listen
to me next time...
On Thu, Jul 15, 2010 at 7:43 PM, Nathaniel Watkins
nwatk...@garrettcounty.org wrote:
Don't let him fool you - you'll get one of those later as well :)
-Original Message-
From:
Well - I consider that an implied 'I told you so'
From: Tony Graziano [mailto:tgrazi...@myitdepartment.net]
Sent: Thursday, July 15, 2010 7:46 PM
To: Nathaniel Watkins
Cc: garyluc...@gmail.com; sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] Polycom IP 430 won't register remotely
Nope.
Love you guys.
Don't worry. If it breaks, i'll come back on and say you told me so
myself.
;-)
On Thu, Jul 15, 2010 at 7:50 PM, Nathaniel Watkins
nwatk...@garrettcounty.org wrote:
Well – I consider that an implied ‘I told you so’
*From:* Tony Graziano
On Thu, Jul 15, 2010 at 4:16 PM, Tony Graziano tgrazi...@myitdepartment.net
wrote:
Its not an issue. There is a JIRA I opened on that. If it were me, nd its
not, I would have bumped to 4.0.4 before going to 4.2.
Thanks for the reply. Just curious...why do you recommend going to 4.0.4
first?
I've been reading through the documentation and threads trying to find the
recommended Polycom SoundPoint SIP Applicaton and BootROM version. I know
that there's a bug with the 3.2.3 SIP Application, so I'll stay away from
that. However, I've seen some recommendations to use 3.2.2 and some to use
Use the latest BootROM
Use 3.2.2 if you don't need SLA
Use 3.1.3RevC if you need SLA
On 07/15/2010 08:20 PM, Tim Byng wrote:
I've been reading through the documentation and threads trying to find
the recommended Polycom SoundPoint SIP Applicaton and BootROM version.
I know that there's a bug
On Thu, Jul 15, 2010 at 9:21 PM, Josh Patten jpat...@co.brazos.tx.uswrote:
Use the latest BootROM
Use 3.2.2 if you don't need SLA
Use 3.1.3RevC if you need SLA
Thanks for the response Josh. I don't need SLA.
I just found the following from Tony: Don't uise above 3.1.3RevC on a
remote
On 7/15/2010 8:40 PM, Tim Byng wrote:
On Thu, Jul 15, 2010 at 9:21 PM, Josh Patten jpat...@co.brazos.tx.us
mailto:jpat...@co.brazos.tx.us wrote:
Use the latest BootROM
Use 3.2.2 if you don't need SLA
Use 3.1.3RevC if you need SLA
Thanks for the response Josh. I don't need SLA.
On 7/15/2010 8:21 PM, Josh Patten wrote:
Use the latest BootROM
Use 3.2.2 if you don't need SLA
Use 3.1.3RevC if you need SLA
Isn't that backwards? I thought 3.2.2 was the latest that let SLA work
but also introduced the bug with phones over circuits with latency.
Maybe I'm mixing up my
3.2.2 has the SLA bug
3.2.3 has the latency bug
3.1.3RevC has an issue with EHS adapters for wireless headsets where if
a monitored extension is ringing then it signals the headset to ring as
well.
3.1.3RevC is stable but is missing my favorite feature: ANI lookup (AKA
reverse caller ID).
I
2010/7/15 Miguel Jesús Díaz Macedo miguel.diazmac...@gmail.com:
How can I terminate SIP registrations manually?, I have restarted de
registar service, but I still see the registrations as not expired
if the registrar is down, and you're careful, and you're not in HA
setup, you can edit
in the event there are schema changes.
On Thu, Jul 15, 2010 at 8:54 PM, Tim Byng t...@missioninc.com wrote:
On Thu, Jul 15, 2010 at 4:16 PM, Tony Graziano
tgrazi...@myitdepartment.net wrote:
Its not an issue. There is a JIRA I opened on that. If it were me, nd its
not, I would have bumped
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