http://track.sipfoundry.org/browse/XX-10120
--
~~
Tony Graziano, Manager
Telephone: 434.984.8430
sip: tgrazi...@voice.myitdepartment.net
Fax: 434.465.6833
~~
Linked-In Profile:
http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4
Ask about our Internet Fax
http://track.sipfoundry.org/browse/XX-10121
--
~~
Tony Graziano, Manager
Telephone: 434.984.8430
sip: tgrazi...@voice.myitdepartment.net
Fax: 434.465.6833
~~
Linked-In Profile:
http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4
Ask about our Internet Fax
http://track.sipfoundry.org/browse/XX-10122
Relates to error message handling when voicemail is not enabled for the
user and they try to check it.
Also relates to being able to change the IVR message for the announced
transfer process or the ability to skip it by admin option.
--
Where I am lost is that Ekiga work with the FQDN of the asterisk server but
not with the FQDN of my SipXecs server in my simple subnet test network.
When I had the problem I though it was Ekiga because it was working well
with X-lite but now I don't know if it's not the server because Ekiga work
2012/4/19 Simon Brûlé sbr...@360-innovations.com:
Where I am lost is that Ekiga work with the FQDN of the asterisk server but
not with the FQDN of my SipXecs server in my simple subnet test network.
When I had the problem I though it was Ekiga because it was working well
with X-lite but now I
Then I suggest you fix your DNS or set the account appropriately.
I just configured EKIGA in 1 minute on ym LAN, it works.
Name: whatever
registrar: sipx-hostname
User: subscriber, ie. 200
Authentication user: subscriber, ie. 200
Password: sipx user sip password, not PIN
My domain has an alias
In my System -- Domain I got the hostname of my server as an alias.
The Ekiga config is the following:
Name:3014
Registrar:voiptest.netappsid.local(the hostname of my server)
User:3014
Authentification User:3014
Password:The generated sip password
Timeout:3600
and it give me a Transport Error
It does noone any good when you talk about that. It sounds like a network
or PC related issue (i.e. local firewall settings, etc.).
It's still an EKIGA error and it points back to your system(s).
Transport error n/a Local_BadTransportAddress
Sounds very reasonable, I have quite a few extensions without vm that would
benefit.
Voted +1.
On Thu, Apr 19, 2012 at 8:47 AM, Tony Graziano tgrazi...@myitdepartment.net
wrote:
http://track.sipfoundry.org/browse/XX-10122
Relates to error message handling when voicemail is not enabled for
Ok thank you i will check with the Ekiga forum.
2012/4/19 Tony Graziano tgrazi...@myitdepartment.net
It does noone any good when you talk about that. It sounds like a network
or PC related issue (i.e. local firewall settings, etc.).
It's still an EKIGA error and it points back to your
In the option of ekiga i don't have a Stun Server option but when I go in
the Ekiga configuration with the Configuration Editor tool on Ubuntu 11.10
i see the stun server option and it's disabled.
2012/4/19 Tony Graziano tgrazi...@myitdepartment.net
perhaps you have enabled STUN (not needed
My Ekiga gave me the error that he cannot discover the
network automatically so I would need to configure it manually and they
gave me a page with instruction to forward port from the router (I imagine
that is for people using Ekiga.net account). Can the part with Ekiga not
being able to discover
I dont use an ekiga account/call out account. I skip all that and just
create a sip account (only).
2012/4/19 Simon Brûlé sbr...@360-innovations.com
My Ekiga gave me the error that he cannot discover the
network automatically so I would need to configure it manually and they
gave me a page
Ok thanks for the help its really appreciated. I will check with the people
from Ekiga see if they can help me solve my problem.
2012/4/19 Tony Graziano tgrazi...@myitdepartment.net
I dont use an ekiga account/call out account. I skip all that and just
create a sip account (only).
2012/4/19
Tony,
I modified my profile voip default. It didn't accept some of the
command, and the configuration is now:
/profile voip default
codec 1 g711ulaw64k rx-length 20 tx-length 20
codec 2 transparent-clearmode rx-length 20 tx-length 20
no dtmf-relay
rtp traffic-class local-default
no
Since when have I had to set up a manually configured sip phone until now?
You are still thinking like this is old hat to everyone using the system.
Thank you for making my point.
Even if someone has been a traditional phone guy for a long time, if he
has never messed with SIP at all, how
your statements show you as out of order. Read the wiki should be first,
which is what leads to a string of a dozen plus messages. Stop making the
wrong points.
Good luck.
On Thu, Apr 19, 2012 at 12:09 PM, Stiles Watson wat...@datatek-net.comwrote:
Since when have I had to set up a manually
no *codec 2 g711alaw64k rx-length 20 tx-length 20*
On Thu, Apr 19, 2012 at 11:27 AM, Jesse Becker beck...@sunyulster.eduwrote:
Tony,
I modified my profile voip default. It didn't accept some of the command,
and the configuration is now:
*
profile voip default
codec 1 g711ulaw64k
Looks like that part of the wiki is wrong then, I've changed to the wiki
entry to show that the password used should be the SIP password.
On Thu, Apr 19, 2012 at 12:09 PM, Stiles Watson wat...@datatek-net.comwrote:
Since when have I had to set up a manually configured sip phone until now?
Folks -
Just got the Samsung Galaxy Player 4.0 and... YOW! Works really well as a
SIP handset so far with the CSipSimple app. Setup and registration with
SipX? No problem. High noise environment? No problem. Hand-off between
AP's? No problem. Clarity of the call? No problem.
My head is suddenly
no, its not wrong, I have changed it back...
Only the Bria 3.x is capable of being auto provisioned. The iPhone, iPad
and Android editions do not have this feature from Counterpath.
Starting Bria
When launching Bria 3.x you have to provide Username, Password and
Provisioning server.
For
Out of context, I now see it more clearly. It's not so clear in the wiki
that the Startup section is intended for the provisioning mode.
So, I've changed the section title to 'Starting Bria in provisioned mode'.
Now it's way clear.
On Thu, Apr 19, 2012 at 12:30 PM, Tony Graziano
Tony,
I had guessed the no as I thought it would be similar to Cisco. The issue
was that you have to omit the number (ie. 2) for it to work.
Any ideas on why it still isn't does t.38? Do I need to make the same
modifications you sent me for the 4424 to the SN4950 (T1 gateway) ?
Thanks,
Jes
On
No. I thought I sent you a 4960 config some time ago. The most recent one I
put on the wiki does work in conjunction with the 4424 profile I already
gave you though.
I'm sure if it doesnt work after comparing, patton can fix with you though.
On Thu, Apr 19, 2012 at 12:38 PM, Becker, Jesse
it also says for 3.x and earlier on the page it says 3.x is provisionable,
not the mobile apps.
On Thu, Apr 19, 2012 at 12:37 PM, Philippe Laurent p...@ideos.com wrote:
Out of context, I now see it more clearly. It's not so clear in the wiki
that the Startup section is intended for the
Tony,
My PRI config matches yours. I will get in touch with Patton to see if
they can help me get this resolved.
Thanks,
Jes
On Thu, Apr 19, 2012 at 12:42 PM, Tony Graziano
tgrazi...@myitdepartment.net wrote:
No. I thought I sent you a 4960 config some time ago. The most recent one
I put
I will send you an fxs config I use all the time offline/
On Thu, Apr 19, 2012 at 12:51 PM, Becker, Jesse beck...@sunyulster.eduwrote:
Tony,
My PRI config matches yours. I will get in touch with Patton to see if
they can help me get this resolved.
Thanks,
Jes
On Thu, Apr 19, 2012 at
Tony,
Thanks again for your help. I just spoke to Patton... great support by the
way. Real person even. They stated there are a lot of detection problems
with the 4425 and recommended updating to latest firmware on both devices.
I am going to do the FXS now and will do the PRI gateway after
On 04/18/2012 06:15 PM, Tony Graziano wrote:
You shouldn't invite violence. I have been known to clobber people
with iPads.
Why am I not surprised by this?
I think the wiki could be clearer, but really I think you are the only
one to make the leap from Bria is bria is bria is bria. It ain't,
I don't have any answer yet concerning my issue so I downloaded Jitsi
instead and it's working pretty well absolutely no problem configuring it
and working on the first try :D .
Now I am going to work on my No Sound problem when I am connected on the
SipXecs server from an other subnet on the
I posted a while back about all of the phones deregistering at the same time,
and was told that it was because I only had 1 GB of ram in my server
A week ago, I upgraded the hardware to this:
HP ProLiant ML110 G7 664723-S01 4U Micro Tower Entry-level Server - 1 x Core i3
i3-2120 3.3GHz - 4 GB
On Thu, Apr 19, 2012 at 1:06 PM, Stiles Watson wat...@datatek-net.comwrote:
On 04/18/2012 06:15 PM, Tony Graziano wrote:
You shouldn't invite violence. I have been known to clobber people with
iPads.
Why am I not surprised by this?
Um that was a cartoon reference dude, you take things
It is open to the Internet (port 5060 forwarded at your firewall)?
If so, look at your log sizes for the day prior to your issue and compare
them to today (roxy and registrar logs).
If they are considerable larger today it might be your were the subject a a
DOS attack, inspeacting the larger
yea, they have that new 3.6 unit coming now too that is only $149 retail.
why have hard phones? :-)
mike
On Thu, Apr 19, 2012 at 12:24 PM, Philippe Laurent p...@ideos.com wrote:
Folks -
Just got the Samsung Galaxy Player 4.0 and... YOW! Works really well as a
SIP handset so far with the
I've actually just spent the last few hours digging through the Android SDK
to figure out how to create an Android App wrapper around my internal web
apps, just as I've done with the iOS platform here.
No better way to get distracted and kill the day's schedule than with a
reasonably priced
Hi, I know I already posted something very similiar to this problem but I
haven't found a solution to it so here i am reposting my problem but with
more precision this time.
I have a softphone (Jitis) on a Ubuntu 11.10 installation connected to the
network of the company.
I have a router Linksys
hahaha... sounds like my life :-)
except i don't have to write the code, just dream this stuff up and our
programming wizards figure it out.
On Thu, Apr 19, 2012 at 2:09 PM, Philippe Laurent p...@ideos.com wrote:
I've actually just spent the last few hours digging through the Android
SDK to
Does anyone have an good or bad experience with Microtech FaxFinder
Appliance, either SIP or analog version to share. I'm looking at using it
for an outbound fax solution for a customer, and looking for any experiences
others might have has with this product.
Thanks in advance for any
On 4/19/2012 2:37 PM, Simon Brûlé wrote:
Hi, I know I already posted something very similiar to this problem
but I haven't found a solution to it so here i am reposting my problem
but with more precision this time.
I have a softphone (Jitis) on a Ubuntu 11.10 installation connected to
the
On 4/19/2012 2:34 PM, Sven Evensen wrote:
A customer wants all external calls before 8am to go straight to VM
while the internal calls should ring. Is this possible today somehow
or does anyone know if this is in the roadmap? They are on 4.4 sipx
Have the external calls go to a phantom user
I added 192.168.175.0/24 to the intranet subnet and I still have the same
problem.
2012/4/19 Gerald Drouillard gerryl...@drouillard.ca
On 4/19/2012 2:37 PM, Simon Brûlé wrote:
Hi, I know I already posted something very similiar to this problem but I
haven't found a solution to it so here i
Do you mean MultiTech?
I haven't used their SIP stuff but I did use their analog versions and they
are great.
On Thu, Apr 19, 2012 at 2:45 PM, Todd Hodgen thod...@frontier.com wrote:
Does anyone have an good or bad experience with Microtech FaxFinder
Appliance, either SIP or analog version
I have several of the analogs installed, good devices.
On Thu, Apr 19, 2012 at 2:58 PM, Michael Picher mpic...@ezuce.com wrote:
Do you mean MultiTech?
I haven't used their SIP stuff but I did use their analog versions and
they are great.
On Thu, Apr 19, 2012 at 2:45 PM, Todd Hodgen
Hi Todd!
Yep, we just got a batch like that. Would you mind providing a link to the
utility needed to perform the downgrade to 3.2?
Thanks.
- Original Message -
From: Todd Hodgen thod...@frontier.com
To: Discussion list for users of sipXecs software
sipx-users@list.sipfoundry.org
On 4/19/2012 2:58 PM, Simon Brûlé wrote:
I added 192.168.175.0/24 http://192.168.175.0/24 to the intranet
subnet and I still have the same problem.
2012/4/19 Gerald Drouillard gerryl...@drouillard.ca
mailto:gerryl...@drouillard.ca
On 4/19/2012 2:37 PM, Simon Brûlé wrote:
Hi, I know
Yep, my bad. Multitech it is.
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Michael Picher
Sent: Thursday, April 19, 2012 11:59 AM
To: Discussion list for users of sipXecs software
Subject: Re: [sipx-users] Experience with Microtech
Great, thanks Philippe.
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Philippe
Laurent
Sent: Thursday, April 19, 2012 12:04 PM
To: Discussion list for users of sipXecs software
Subject: Re: [sipx-users] Experience with Microtech
Won't a format file system under the admin menu force it to pull the
bootrom and firmware from the SipX server? Would that be easier than using
a special tool?
On Thu, Apr 19, 2012 at 3:07 PM, andrewpit...@comcast.net wrote:
Hi Todd!
Yep, we just got a batch like that. Would you mind
How can I do a capture with wireshark on the SipXecs server?
About the ALG you think that the other Router that give the DHCP to my
Laptop and the Wan adresse of my router would have the Sip ALG activate?
2012/4/19 Gerald Drouillard gerryl...@drouillard.ca
On 4/19/2012 2:58 PM, Simon Brûlé
I received this link from a Polycom Product Manager. It has all of the
information related to the new UC versions of the Polycom firmware. Version
3.3.x and above they are referring to it as UC.
I don't believe it works that way, and they speak to that in the
documentation. There is a utility to upgrade, and a utility to downgrade.
I don't know if this is a different form of format, or what, but they
specifically speak to using these utilities once a phone is moved over the
line.
On 4/19/2012 3:25 PM, Simon Brûlé wrote:
How can I do a capture with wireshark on the SipXecs server?
If you google a little you will find it.
About the ALG you think that the other Router that give the DHCP to my
Laptop and the Wan adresse of my router would have the Sip ALG activate?
That
I've created a wiki guide within the Polycom section geared towards setting
up the KIRK server with SIPX. It's not written in a very technical style,
but it works nonetheless.
The two dozen KIRK phones that I've got get along well with SIPX.
Thanks for the opportunity to allow me to contribute.
Yeah, I was wondering the same thing myself as I've successfully gone back to
3.1 from 3.2 with a format, but I didn't want to risk making myself some
Polycom bricks myself with these brand new ones. ;)
- Original Message -
From: Todd Hodgen thod...@frontier.com
To: Discussion list
Do you mean multi tech?
On Apr 19, 2012 2:47 PM, Todd Hodgen thod...@frontier.com wrote:
Does anyone have an good or bad experience with Microtech FaxFinder
Appliance, either SIP or analog version to share. I’m looking at using it
for an outbound fax solution for a customer, and looking for
My Computer is connected in the Lan of the company and my E2500 is
connected in this Lan too. My Server SipXecs and my hardphone are on the
E2500.
So there is an other router between my computer and the router that have my
Server connected on it.
For the Wireshark part when I answer the
I have successfully moved between firmware and bootroms with the format
file system without issue using the format options. If I end up getting a
newer phone I may give it a shot to see if that is still the case after
3.3. Worse case, I RMA the new phone.
On Thu, Apr 19, 2012 at 3:43 PM,
As i dig more and more in Wireshark i came to the conclusion that the
Wireshark information that I just sent you is pretty much useless as I now
see it. I will keep looking for some piece of information that could help.
Thanks.
2012/4/19 Simon Brûlé sbr...@360-innovations.com
My Computer is
Yes, and that is fully supported by Polycom.
What is not supported is moving from pre-3.3 to post 3.3 without using the
update tool. It's in their documentation. As a small business owner, I
don't have the luxury of an employer to replace a phone I damage, so I use
an abundance of
There doesn’t seem to be any RTP from the .136 device in this capture. Have
you check at that device to see if it is sending RTP, is there a Firewall
blocking RTP from that direction?
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of
If there is NAT between your PC and the sipx server then:
1. The local firewall to your PC needs to have any SIP helper or
Application Layer Gateway turned off.
2. At your firewall where sipx is the SIP helper or Application Layer
Gateway needs to be turned off AND the NAT type for the outbound
So your saying the problem may come from the router I have (Linksys E2500)
because it's not doing the symmetrical Nat so the RTP is getting lost?
2012/4/19 Tony Graziano tgrazi...@myitdepartment.net
If there is NAT between your PC and the sipx server then:
1. The local firewall to your PC
Most likely if that is the router sitting in front of sipx, yes. It's a
prerequisite for media relay to function.
1. Server behind nat has to be enabled.
2. Support remote workers needs to be enabled (sip trunking does not need
to be enabled for remote workers, media relay does).
3. NAT for
or could be stun too if the media relay's just off in left field.
i would setup for static outside address if you haven't.
mike
On Thu, Apr 19, 2012 at 4:59 PM, Tony Graziano tgrazi...@myitdepartment.net
wrote:
Most likely if that is the router sitting in front of sipx, yes. It's a
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