.. to add some more data on my own question ...
I logged on on the freeswitch CLI when I am about to leave the voice mail.
I can see the record app being kicked out with the filename and then 300
500 10 as params passed to the record app
-- snip --
Command Execute
record(/var/sipxdata/media
Thanks Scott, Dale and the SIP RFCs. I now understand how MWI works.
I have tracked this problem down to a point where I think it is a Snom bug. The
phones aren't always sending a new SUBSCRIBE request when the old one expires.
There is some bug in their countdown timer because, sometimes, the t
On Wed, 2010-02-17 at 12:56 -0800, Pizza Napoletana wrote:
> Using sipx 4.1.6.
>
> My snom 870 phones seem to get MWI NOTIFY messages correctly for a few
> hours after the phones are rebooted. But after those few hours, when a
> message is left, no NOTIFY messages are sent out from sipx
> (per /va
On Wed, 2010-02-17 at 12:56 -0800, Pizza Napoletana wrote:
> Using sipx 4.1.6.
>
> My snom 870 phones seem to get MWI NOTIFY messages correctly for a few
> hours after the phones are rebooted. But after those few hours, when a
> message is left, no NOTIFY messages are sent out from sipx
> (per /va
Using sipx 4.1.6.
My snom 870 phones seem to get MWI NOTIFY messages correctly for a few hours
after the phones are rebooted. But after those few hours, when a message is
left, no NOTIFY messages are sent out from sipx (per /var/log/sipxlogs/*). And
if I reboot the phone again, everything is fi
On Feb 17, 2010, at 7:53 AM, Peter Fowler wrote:
> Can you provide more information on the case where you turned off
> notification.
> In what way did sipXivr get confused. What specific behavior or logs do
> you see?
Yesterday, all it took was to turn on imap sync for a user and then turn it off
sipXconfig (4.0.4-017289 2009-11-19T18:15:29 ecs-centos5)
- Original Message -
From: "Robert Joly"
To: "Francis Tinio" , "sipx-users"
Sent: Wednesday, February 17, 2010 2:23:07 PM GMT -05:00 US/Canada Eastern
Subject: RE: [sipx-users] calls not disconnecting
> I just noticed when loo
> I just noticed when looking at the call details records that
> I have a call in progress, and it's been running for almost
> an hour now. The call was the one I made earlier that was
> not answered, and I have already hung up on my end as well as
> the other end. So how come it's still runn
yea I did. What I meant was it reflected as in progress even in the CDR of my
ITSP, so I assumed that the call was actually in progress and not a misreport
as what is said on the right hand side of the CDR of sipx.
Sorry for the confusion. I'll check my ITSP, as the calls should not reflect
o
Did you even read my response?
Josh Patten
Assistant Network Administrator
Brazos County IT Dept.
(979) 361-4676
On 2/17/2010 1:15 PM, Francis Tinio wrote:
> yup it says that it's in progress, when it should have been disconnected
> already. I had to reboot the sipx server just to stop it.
>
>
yup it says that it's in progress, when it should have been disconnected
already. I had to reboot the sipx server just to stop it.
- Original Message -
From: "Josh Patten"
To: sipx-users@list.sipfoundry.org
Sent: Wednesday, February 17, 2010 2:12:21 PM GMT -05:00 US/Canada Eastern
Subj
Read the entirety of the CDR page including all text on the right side.
Josh Patten
Assistant Network Administrator
Brazos County IT Dept.
(979) 361-4676
On 2/17/2010 1:11 PM, Francis Tinio wrote:
> I just noticed when looking at the call details records that I have a call in
> progress, and it
I just noticed when looking at the call details records that I have a call in
progress, and it's been running for almost an hour now. The call was the one I
made earlier that was not answered, and I have already hung up on my end as
well as the other end. So how come it's still running as a ca
IP500 maxes at 2.1.3
while the IP 500 has a configuration option in sipX, it is not
recommended for use.
http://wiki.sipfoundry.org/display/xecsuser/Polycom
Josh Patten
Assistant Network Administrator
Brazos County IT Dept.
(979) 361-4676
On 2/17/2010 11:50 AM, Tony Graziano wrote:
> Use a cu
Use a current firmware...
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.myitd
If it was endian that's causing this, then shouldn't the softphones not work as
well? How come the issue is only with polycom, and only for outgoing calls.
incoming calls have 2 way audio.
- Original Message -
From: "Tony Graziano"
To: fti...@toqen.com, sipx-users@list.sipfoundry.org
My experience has been that it will work on again/off again with endian,
hence my suggestion to not use it. IF you used a pfsense firewall and had
the same issues I would be surprised. Also realize that the firmware you are
on is problematic (2.2), whereas you should be ysing 3.1.3RevC.
With pfsen
You assume the gateway will (at&t) will handle moh, process and accept
REFER, etc. They won't and an SBC of some nature needs to anchor the media
accordingly. I would not recommend a dynamic rule to handle media in this
fashion, the guidance for sipXbridge has always been to use a set rule, not
a d
Hi.
I thought I was out of the loop last night when I got the polycoms to register.
However, it seems that incoming calls work fine but outgoing calls do not have
any audio.
This only happens on the polycom ip500. The Bria softphones and x-lites all
work without any issues for both incoming
Just a few updates:
> First there is a problem with when the port is opened.
This can be resolved with static NAT.
> This is possible if the firewall performs the ALG
> function by opening the port within the SDP and
> creating a DNAT rule to forward the traffic to the
> phone.
AND if the firew
Tony Graziano said,
"They are supposed to be the same. The phone does not matter here.
Sipxbridge manages that. Your vyaTta router is not doing symmetric nat."
Ok, I see that now. I traced the packets on the public side and indeed the
firewall is rewriting the port number. So if I can change my f
On Feb 17, 2010, at 8:55 AM, Scott Lawrence wrote:
> On Tue, 2010-02-16 at 18:36 -0600, Eric Varsanyi wrote:
>>
>> My question is: shouldn't *all* outbound calls through a gateway have
>> their 'From' remapped based on the gateway settings?
>
> The current design intent is that they should not.
On Tue, 2010-02-16 at 18:36 -0600, Eric Varsanyi wrote:
>
> My question is: shouldn't *all* outbound calls through a gateway have
> their 'From' remapped based on the gateway settings?
The current design intent is that they should not.
This has been discussed often enough that I'm going to creat
On 2/16/10 4:05 PM, Tony Graziano wrote:
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.myitde
Hello list,
I have the following issue with the voicemail: every now and then, when
somebody soft spoken leaves a voice mail they get interrupted by the
silence detection threshold and unfortunately they don’t bother to
re-record and speak louder instead they hang up. If there where for me I
Hi Eric
Thanks for the pointer.
I've set 'escape display name' to 'yes' and removed the CLD prefix I had added
("T: ") and this is now working. Perhaps it was not escaping the : or space
char from the CLD prefix I had added.
Cheers
Jesse
On 17/02/2010, at 10:35 AM, Eric Varsanyi wrote:
> T
Thanks, Tony.
Can someone running 4.1.6 confirm that deleting a voicemail message using the
sipx GUI indeed removes the IMAP synchronized email notification?
In my setup, I am noticing that deleting from the phone interface removes the
email, but deleting from the browser GUI does not.
On Feb 1
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