On Feb 16, 2010 8:32pm, Eric Varsanyi wrote:
You did not mention what version of sipx you are running. If it is
4.0.4 then I am very sure you have a configuration problem because
this is something that is surely working.
The From header looks like you did not set your caller ID setting for
I never could use endian because it did not do symettric nat properly.
I've detailed ports here already:
http://blog.myitdepartment.net/?p=37
On Tue, Feb 16, 2010 at 8:26 PM, Francis Tinio wrote:
> btw, what ports are needed open in the firewall again?
>
> I opened all ports with the new endia
btw, what ports are needed open in the firewall again?
I opened all ports with the new endian, it might be that also, since it has to
be 1:1 NAT right?
Thanks
On Feb 16, 2010, at 6:30 PM, Tony Graziano wrote:
> Moh needs to be disabled on the phone.
>
>
> Tony Gr
>
> You did not mention what version of sipx you are running. If it i s
> 4.0.4 then I am very sure you have a configuration problem because
> this is something that is surely working.
>
> The From header looks like you did not set your caller ID setting for
> the gateway.
>
> Ranga
I'll past
OK this is really weird. Now all my installations exhibit the same behavior.
The existing Bria softphones that were working before now are not giving any
dial tone or MOH or any audioalthough when a call is placed or received
there is still 2-way communication
..what's worse is now th
On Tue, Feb 16, 2010 at 7:55 PM, Tony Graziano
wrote:
> Well, the overall issue is understanding what AT&T is doing. A siptrace is
> meaningful. It's as if they are bypassing sipxbridge and sipxbridge manages
> REFER and they (AT&T) do not support it. Those functions (transfers to
> others, voicem
On Tue, Feb 16, 2010 at 7:50 PM, Eric Varsanyi wrote:
> On Feb 16, 2010, at 6:40 PM, M. Ranganathan wrote:
>
>> On Tue, Feb 16, 2010 at 7:36 PM, Eric Varsanyi wrote:
>>> Making calls from a Patton FXS line my ITSP (voip.ms) was giving me reorder
>>> tones (as audio) to some numbers due to my ins
Well, the overall issue is understanding what AT&T is doing. A siptrace is
meaningful. It's as if they are bypassing sipxbridge and sipxbridge manages
REFER and they (AT&T) do not support it. Those functions (transfers to
others, voicemail or MOH ) use REFER. AT&T should only be talking to
sipxbrid
On Feb 16, 2010, at 6:40 PM, M. Ranganathan wrote:
> On Tue, Feb 16, 2010 at 7:36 PM, Eric Varsanyi wrote:
>> Making calls from a Patton FXS line my ITSP (voip.ms) was giving me reorder
>> tones (as audio) to some numbers due to my inside CID being sent to the ITSP.
>>
>> The inside device caus
Yes, Route is set to sipXbridge-1.
So close but yet so far!
Looks like an inbound call from my cell phone to my desk phone hangs up if I
put myself on hold. Same basic issue as with a transfer.
I am going through piece by piece testing one change at a time.
Andrew
_
From: Ton
On Tue, Feb 16, 2010 at 7:36 PM, Eric Varsanyi wrote:
> Making calls from a Patton FXS line my ITSP (voip.ms) was giving me reorder
> tones (as audio) to some numbers due to my inside CID being sent to the ITSP.
>
> The inside device causing trouble was sending a From with its IP address
> rathe
Making calls from a Patton FXS line my ITSP (voip.ms) was giving me reorder
tones (as audio) to some numbers due to my inside CID being sent to the ITSP.
The inside device causing trouble was sending a From with its IP address rather
than the sipx machines domain name, I fixed this and all is we
I see the AT&T flex Ip is a local gateway. When you created the siptrunk,
did you choose sipxbridge as the default route for the trunk?
On Tue, Feb 16, 2010 at 7:19 PM, Tony Graziano wrote:
> forget the codec statement, my mind was elsewhere.
>
>
> On Tue, Feb 16, 2010 at 7:16 PM, Tony Graziano
On Tue, Feb 16, 2010 at 7:22 PM, Andrew Cotter <
andrew.cot...@somersetcapital.com> wrote:
> 1. Did you make sure Internet calling was disabled?
> Disabled
>
>
> 2. When you created the siptrunk, did you use the ATT template?
> Yes - ATT template
> One thing they have not
1. Did you make sure Internet calling was disabled?
Disabled
2. When you created the siptrunk, did you use the ATT template?
Yes - ATT template
One thing they have not giving me (so I put the SIP server IP in) is
the "ITSP server domain name" tech claims she does n
forget the codec statement, my mind was elsewhere.
On Tue, Feb 16, 2010 at 7:16 PM, Tony Graziano wrote:
> You might also check the codec order preference on your phones. G711ulaw
> and G711alaw should be first.
>
>
> On Tue, Feb 16, 2010 at 7:08 PM, Tony Graziano <
> tgrazi...@myitdepartment.ne
You might also check the codec order preference on your phones. G711ulaw and
G711alaw should be first.
On Tue, Feb 16, 2010 at 7:08 PM, Tony Graziano wrote:
> 1. Did you make sure Internet calling was disabled?
> 2. When you created the siptrunk, did you use the ATT template?
> 3. Is AT&T behind
On Tue, Feb 16, 2010 at 7:05 PM, Eric Varsanyi wrote:
> The firmware for the SP3102 has a bug (longstanding) that makes it put out
> corrupted (non sip compliant) headers. It used to be this crashed Freeswitch
> with a SEGV but that's been fixed post 4.0.4.
>
> This *might* be your issue, you can
1. Did you make sure Internet calling was disabled?
2. When you created the siptrunk, did you use the ATT template?
3. Is AT&T behind your firewall or in front?
4. Is the address of your sipx server added as a domain alias?
5. Under Internet calling, did you ensure only your subnets are listed unde
The firmware for the SP3102 has a bug (longstanding) that makes it put out
corrupted (non sip compliant) headers. It used to be this crashed Freeswitch
with a SEGV but that's been fixed post 4.0.4.
This *might* be your issue, you can test by putting in a manual caller id in
the SPA configuratio
To possibly rule out the Polycom firmware/bootrom issue, I created a phantom
user and vmail box. Calling inbound to the AA and selecting the extension
gives me the same dead air issue.
btw - I think I said this before in previous emails on the subject, but the
transfer issue is not a problem o
I have enabled MOH on the sipXbridge-1 and the only parameter on the phones
I can find is "musicOnHold.uri" which is blank in two places. I am looking
at the group the phones are in and under "Lines | Registration" as well as
"Phones | SIP".
I added in the SBC SIP config "Public Port" to be 50
1. Don't think so.
2. IMAP server must support "searches" within headers.
3. Still being worked on. The tracker shows issues with the phone channg
settings and wiping the imap config. You should consider checking the
tracker and opening issues accordingly, making sure by checking you are not
oipeni
I have been playing with voicemail<->imap integration in 4.1.6. I have a few
questions.
1. If I delete an email notification, the corresponding voicemail stored in
/var/sipxdata/mediaserver/data/mailstore is automatically deleted. That's good,
but if I delete a voicemail via sipx GUI, the corre
ITSP is the Internet Telephony Service Provider (carrier).
Are you using one?
On Tue, Feb 16, 2010 at 6:16 PM, Francis Tinio wrote:
> I have sipxbridge setup, Internal SBC on local. ITSP?
>
>
>
> On Feb 16, 2010, at 6:11 PM, Tony Graziano wrote:
>
> Ah. Are you using an ITSP? sipXbridge?
>
> On
"If" it were me, I'd be using bootrom 4.2.(whatever) but I don't think it's
the bootrom.I've been wrong a couple of times.
I would make sure MOH is enabled on sipXbridge and that the MOH field is
blanked out in the phone via sipxconfig and resend the profiles.
If that does not work, I would do a
I have sipxbridge setup, Internal SBC on local. ITSP?
On Feb 16, 2010, at 6:11 PM, Tony Graziano wrote:
> Ah. Are you using an ITSP? sipXbridge?
>
> On Tue, Feb 16, 2010 at 6:10 PM, Francis Tinio wrote:
> I already wiped the polycom phone. Even when I try to call the phone, the
> call goes
Polycoms 430 and 550. Testing on the 550.
bootrom is 4.1.4 and firmware is 3.1.3RevC split.
Andrew
_
From: Tony Graziano [mailto:tgrazi...@myitdepartment.net]
Sent: Tuesday, February 16, 2010 6:13 PM
To: Andrew Cotter
Cc: mpic...@cmctechgroup.com; sipx-users@list.sipfoundry.org
Su
What phone are you using? If Polycom, what bootrom and firmware?
On Tue, Feb 16, 2010 at 6:11 PM, Andrew Cotter <
andrew.cot...@somersetcapital.com> wrote:
> Well... Got AT&T on the phone and they made the port change. I dropped
> the
> unmanged gateway and can once again make calls in and out.
Well... Got AT&T on the phone and they made the port change. I dropped the
unmanged gateway and can once again make calls in and out. AT&T tech
confirmed UDP traffic on port 5080.
Transfers still don't work. Any thoughts?
Andrew
> -Original Message-
> From: Tony Graziano [mailto:tg
On Tue, Feb 16, 2010 at 5:34 PM, Burden, Mike wrote:
> Actually, it fixed it on X-Lite, but not Bria Bria fails to
> register.
>
> The PC is on a subnet that doesn't have access to a nameserver that
> lists an SRV record for lynk.com, so I'm not even sure how X-Lite found
> the sipXecs server
Actually, it fixed it on X-Lite, but not Bria Bria fails to
register.
The PC is on a subnet that doesn't have access to a nameserver that
lists an SRV record for lynk.com, so I'm not even sure how X-Lite found
the sipXecs server...Isn't that what the proxy setting is supposed
to be for?
You should never register at the ip address. If you are using bria pro, let
sipx configure the phone.
Tony Graziano, Manager
Telephone: 434.984.8430
Fax: 434.984.8431
Email: tgrazi...@myitdepartment.net
LAN/Telephony/Security and Control Systems Helpdesk:
Telephone: 4
That fixed it! Thanks!!
Mike Burden
Lynk Systems, Inc
e-mail: m...@lynk.com
Phone: 616-532-4985
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Burden,
Mike
Sent: Tuesday, February 16, 2010 5:12 PM
To: Tony
Thanks! I'll give that a try.
Mike Burden
Lynk Systems, Inc
e-mail: m...@lynk.com
Phone: 616-532-4985
-Original Message-
From: Tony Graziano [mailto:tgrazi...@myitdepartment.net]
Sent: Tuesday, February 16, 2010 5:12 PM
To: Burden, Mike; sipx-users@list.sipfoundry.org
Subject: Re: [
Therin lies your error. Do not register at the ip, use the domain.
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
Help
Good afternoon,
I have a Polycom IP650 that is on the same subnet as the sipXecs server,
and a soft phone (either X-Lite or Bria - both exhibit the same
problem). Both are configured as extension 12.
The softphone is configure with my extension as the username, "lynk.com"
as the domain, a
They are supposed to be the same. The phone does not matter here. Sipxbridge
manages that. Your vyaTta router is not doing symmetric nat.
Tony Graziano, Manager
Telephone: 434.984.8430
Fax: 434.984.8431
Email: tgrazi...@myitdepartment.net
LAN/Telephony/Security and Co
It is. The callerid setting on the sipura should niot be bellcore though.
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: 434.984.842
From: Tony Graziano
> The vyatta router should do "symmetric port nat". If the vyatta is doing
symettic port nat with all ALG or sip filters off, "IT WILL WORK".
If I understand you correctly this will only work if the outbound and
inbound ports are the same. They currently are not. Here are t
Thanks Eric and Tony
On the SPA-3000 I have enabled "Detect Disconnect Tone" and set the Disconnect
Tone to the one given at the following web page for Australia / Telstra PSTN
lines:
http://www.voip-info.org/wiki/view/Sipura+3000
Namely:
4...@-30,4...@-30;1(.375/.375/1+2)
This has fix
Nothing pretty.
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 Contract Customers:
http://www.myitdepartment.
For anyone following this, an upgrade from eyeBeam to Bria appears to
have fixed the root cause of the problem, and the firewall vendor has a
patch on the way to fix the symptom.
Mike Burden
Lynk Systems, Inc
e-mail: m...@lynk.com
Phone: 616-532-4985
-Original Message-
From: sipx-user
OK. My guess is they don't. AT&T is not quite like some of the other
players out there. I have played with folks like flowroute, junction, etc.
that have interfaces. Nothing like that I have heard or from AT&T.
Any route to take if the 800 pound gorilla won't budge and has to send the
calls to
Your vyatta router does not need a sip filter or their "version" of a sip
filter to do this.
The vyatta router should do "symmetric port nat". If the vyatta is doing
symettic port nat with all ALG or sip filters off, "IT WILL WORK".
http://blog.myitdepartment.net/wp-content/uploads/2009/11/Call-S
I have no idea if they do, "MOST I
ITSP's do".
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 Contract Custom
They have a control panel? That one is news to me. Guess my salesperson is
about to get a call!
Andrew
> -Original Message-
> From: Tony Graziano [mailto:tgrazi...@myitdepartment.net]
> Sent: Tuesday, February 16, 2010 1:36 PM
> To: andrew.cot...@somersetcapital.com;
> mpic...@cmctec
Dear sipx list,
I have the following network configuration:
Polycom VVX1500 at 10.253.0.16
Sipx 4.0.4-017289 at 10.253.0.51
Vyatta firewall 3.4.7 - Public IP 1.2.3.4 (not really), private IP
10.253.0.254
Gateway RNK at 207.2.123.180
A call is established between the Polycom and an ou
On Fri, 2009-11-27 at 09:35 +, Gabor Paller wrote:
> The result means to me that SipX under load requires quite an amount of
> memory (beside CPU load). Is this conclusion correct?
Oh, yes. 1 Gb is the minimum, maybe 512 Mb if there aren't many users.
Dale
_
I sent him the disconnect code for the unit for australia.
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 Con
I had this problem too with an SPA 3102. I could never get the linksys firmware
to properly detect CPC via the normal telco means (battery reversal or loop
drop). On the line I was using I could watch with a storage scope and see the
loop break for about 500ms but the SPA would just ignore it no
Once you get it properly configured it's rock solid. There are so many
options and tweakable settings that you can get lost fairly easily. I
have found that reading the (rather long) manual a couple of times is a
really good thing to do. The biggest issue I had with mine was that
Adtran's DMS-1
Or log into their control panel and set it yourself.
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 Contract
I know that 5080 is the default port for sipXbridge. To make sure my calls
are coming on in 5080, do I need to request that from AT&T?
Andrew
> -Original Message-
> From: Picher, Michael [mailto:mpic...@cmctechgroup.com]
> Sent: Tuesday, February 16, 2010 12:47 PM
> To: Andrew Cotter
Make sure that Internet Calling check box is off and that your calls are
actually coming in on 5080 and not 5060.
> -Original Message-
> From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-
> boun...@list.sipfoundry.org] On Behalf Of Andrew Cotter
> Sent: Tuesday, February 16,
The only way to make Adtran gateways work with sipX is to use
sipXbridge. The newest version of their firmware has partial support for
REFER but attended transfers and directed call pickup don't work. I gave
up on my TA908e and bought an Audiocodes Mediant 1000 at twice the
price. I do miss the
On Tue, Feb 16, 2010 at 10:47 AM, Ken Fulmer <
kenful...@icstechnologysolutions.com> wrote:
> Dale,
>
> That's why we're having to use the ITSP functionality of the sipXbridge.
> Based on price, we'd like to use the Adtran voice gateway but they don't
> handle the SIP REFER messages any better tha
Dale,
That's why we're having to use the ITSP functionality of the sipXbridge. Based
on price, we'd like to use the Adtran voice gateway but they don't handle the
SIP REFER messages any better than the Cisco router.
Any suggestions on an affordable voice gateway that handles SIP REFER messages?
On Tue, Feb 16, 2010 at 10:44 AM, Francis Tinio wrote:
> uhmmm.that'll be a problem, coz the phone was deployed for asterisk
> before...all the settings relating to servers I had them point to the sipx
> server, as is the FTP server. But for user/pass for FTP I have no clue what
> to put so
uhmmm.that'll be a problem, coz the phone was deployed for asterisk
before...all the settings relating to servers I had them point to the sipx
server, as is the FTP server. But for user/pass for FTP I have no clue what to
put so I left them blank. It's not provisioning still :(
On Feb 16
But are you guys talking about if someone had access to the wire and was
sniffing or an attack based on random or sequential guessing?
I don't know the term but like a dictionary attack but using names.
On Tue, 16 Feb 2010 10:08:00 -0500, Dale Worley wrote:
> On Tue, 2010-02-16 at 08:30 -0500, S
On Tue, 2010-02-16 at 09:28 -0600, Ken Fulmer wrote:
> We’ve had some problems with the Polycom phones use of SIP REFER
> messages with Cisco and Adtran voice gateways. Call transfers and
> holds fail due to incompatibilities in the way the Cisco and Adtran
> routers handle these events. My underst
On Tue, Feb 16, 2010 at 10:38 AM, Francis Tinio wrote:
> Public IP in the FTP setting? What about the user/password, or I just
> leave them blank?
>
>
> On Feb 16, 2010, at 10:37 AM, Tony Graziano wrote:
>
>
>
> On Tue, Feb 16, 2010 at 10:34 AM, Francis Tinio wrote:
>
>> Do I need to put anythi
Public IP in the FTP setting? What about the user/password, or I just leave
them blank?
On Feb 16, 2010, at 10:37 AM, Tony Graziano wrote:
>
>
> On Tue, Feb 16, 2010 at 10:34 AM, Francis Tinio wrote:
> Do I need to put anything in the FTP settings in the polycom phone?
>
> Also, the server
On Tue, Feb 16, 2010 at 10:34 AM, Francis Tinio wrote:
> Do I need to put anything in the FTP settings in the polycom phone?
>
> Also, the server is located remotely. and The polycom is on another network
> via NAT. The Bria works fine, but I can't get this to auto provision.
>
>
> On Feb 16, 20
Do I need to put anything in the FTP settings in the polycom phone?
Also, the server is located remotely. and The polycom is on another network via
NAT. The Bria works fine, but I can't get this to auto provision.
On Feb 16, 2010, at 10:33 AM, Tony Graziano wrote:
> l fail to restart the phon
If the phones are local to sipx, your dhcp server settings will send the
phones the proper settings, you need to: 1. create the phone in sipxconfig,
2. add a line, 3. send the profile (it will fail to restart the phone
because it was not registered to begin with, 4. On reboot, the phone will
pickup
We've had some problems with the Polycom phones use of SIP REFER messages
with Cisco and Adtran voice gateways. Call transfers and holds fail due to
incompatibilities in the way the Cisco and Adtran routers handle these
events. My understanding is the routers use a mid-call INVITE rather than
SIP R
Hi.
My sipx server has been working fine with a Bria softphone. Now we would like
to setup our polycom ip500 phones with the server. How do we auto provision
it? Where do I find which TFTP or FTP server to get the config settings?
Thanks
--F
___
s
On Tue, 2010-02-16 at 08:30 -0500, Scott Lawrence wrote:
> On Mon, 2010-02-15 at 21:26 -0600, m...@grounded.net wrote:
>
> > In asterisk, one of the main suggestions for security is to make your
> > SIP user names different than your extensions.
>
> A pointless attempt at 'security through obscur
I will give that a try again. Started down that road and had problems
originally. I'll retrace my steps and see if I can get it to work.
Thanks for pointing that out!
Andrew
> -Original Message-
> From: Tony Graziano [mailto:tgrazi...@myitdepartment.net]
> Sent: Tuesday, February
On Mon, 2010-02-15 at 21:26 -0600, m...@grounded.net wrote:
> In asterisk, one of the main suggestions for security is to make your
> SIP user names different than your extensions.
A pointless attempt at 'security through obscurity'. It's trivial to
get a SIP endpoint to reveal its user name.
Hi all!
I had it working...don't know exactly what i did but now it does not work.
I have:
- 4.0.4
- centos
- snom320 phones. 7.3.14 firmware
- all call features working
I want:
- function keys that displays the states of the various extensions.
blink-incoming, lit up-speaking and
to be able to
Is anybody using the Voiceworks ACD replacement for Sipx 4.x as described here:
http://wiki.voiceworks.pl/display/vwost/VoiceWorks+ACD
I'm intrigued by a freeswitch ACD but I'm not sure how well tested this
solution is. If I'm not mistaken, the Voiceworks ACD CTI app will be part of
4.2 but
You need to use sipxbridge or use another SBC. AT&T is dropping the call
because they don't support REFER. You need to follow the steps for
configuring sipxbridge AND a sip trunk for AT&T, not an unmanaged gateway.
Tony Graziano, Manager
Telephone: 434.984.8430
Fax: 434
The sipura is not sensng the hangup properly, it is misconfigured.
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
Help
On 2010-02-16, at 05:20, James R wrote:
> Sorry for the delayed response! This looks great. I'm working on building
> this in our lab. Is this the core of the future sipX ACD replacement?
> (version 5?)... for some reason I can't find the link to it on the issue
> tracker right now.. wasn'
Hello
I've set up a small SIPX setup at home, to have a play with it really and to
'unify' incoming calls via disparate means (voip and pstn) so they can be
answered on the same set of phones. We're also about to switch to sipX at work.
So, the problem I'm having is that when a call comes in o
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