On Apr 27, 2010, at 5:36 PM, Marcello Manzardo wrote:
> Does anybody know whether the Polycom 321 phones play well with 4.2?
We got a couple of them last week. They auto-configure and do all basic things
(like make / receive / transfer calls, play nice with voicemail, etc.) work
fine with 4.2. B
Running 4.2 from ISO
Every 30 seconds, I get the following message in /var/log/sipxpbx/sipXproxy.log:
"2010-04-28T00:32:14.736830Z":39:HTTP:ERR:pbx.phone.mtv.global.com::B6CFDB90:SipXProxy:"HttpMessage::get[4]
Receiving failed on persistent connection on try 0"
Every 30 seconds, I get the fo
When using 4.2 (ISO install), does anyone else get this email everyday?
/etc/cron.daily/voicemail_clean:
Configuration file not found: '/etc/sipxpbx/voicemail-config'
find: invalid argument `+' to `-mtime'
find: invalid argument `+' to `-mtime'
find: invalid argument `+' to `-mtime'
find: invalid
rol Systems Helpdesk:
> Telephone: 434.984.8426
> Fax: 434.984.8427
>
> Helpdesk Contract Customers:
> http://www.myitdepartment.net/gethelp/
>
> - Original Message -
> From: Pizza Napoletana
> To: Tony Graziano
> Cc: jpat...@co.brazos.tx.us ;
> sipx-users@list.sip
04 2010
> Subject: Re: [sipx-users] Is it OK to edit validusers.xml?
>
> http://wiki.sipfoundry.org/display/xecsuser/PIN+retrieval+tool
>
> can be used to extract PIN numbers from the database.
>
> Pizza Napoletana wrote:
>> I decided to do a fresh ISO install of 4.2, rather than try an up
I decided to do a fresh ISO install of 4.2, rather than try an upgrade from
4.1.6. It is a small installation (about 25 users). So manually recreating
users, phones, gateway, dial plans, etc. wasn't a big deal. But I have a couple
of things I need to copy from the old system and I am not sure ho
We have been using it with dovecot. No issues.
I don't know if there is a wiki page that lists IMAP servers that are known to
work well / not well. If there is, dovecot can be added to the good list.
On Apr 19, 2010, at 1:11 PM, Josh Patten wrote:
> What IMAP server are you using? Your IMAP serv
Version: 4.1.16
Feature: Voicemail notification via email
Request #1
Currently, email notifications are formatted as follows:
From: Voicemail Notification Service
Subject: Voice message from ()
I think it will be more useful if formatting is as follows:
From:
Subject: Voicemail
I have a SIP Trunk related question.
Our ITSP requires that all outgoing calls, except 911 calls, use E.164 dial
format, with a "+" prefix.
Adding a + in the Prefix field of the SIP Trunk Gateway's Dial Plan settings,
adds a + to all calls including 911.
To not have + in the beginning of 911 ca
On Mar 25, 2010, at 7:20 PM, Dale D wrote:
> Will the IMAP integration result in the message being
> deleted from the email box if it is deleted in sipx? I hope
> not. There are many users who simply want to forward all
> their VMs to email and have them deleted from sipx. Is/Will
> there be a w
MWI light is still on you may have to
>> restart the phone.
>>
>> Josh Patten
>> Assistant Network Administrator
>> Brazos County IT Dept.
>> (979) 361-4676
>>
>>
>> On 3/22/2010 12:40 PM, Pizza Napoletana wrote:
>>> Using sipxecs 4
Using sipxecs 4.1.6-018058 2010-02-20T17:33:02 ecs-centos5
Out of about 20 users in this setup, one user has voicemail and user portal
access issues now, but she didn't have those issues last week. No configuration
change / software upgrade was done in the mean time.
When this user logs in via
Oops, I meant to send this to the whole group.
Begin forwarded message:
> From: Pizza Napoletana
> Date: March 10, 2010 12:59:25 PM PST
> To: Jeff Gilmore
> Subject: Re: [sipx-users] Outgoing caller ID name & number?
>
> My ITSP says that there is no way for them take t
ht polycom 550's. They're double the price but so much easier to deal
> with. I now use my Grandstream phones as data center and lobby/door phones.
>
> Pizza Napoletana wrote:
>> Using 4.1.6-018058 2010-02-20T17:33:02 ecs-centos5 ...
>>
>> I set syslog serve
Using 4.1.6-018058 2010-02-20T17:33:02 ecs-centos5 ...
I set syslog server and NTP server parameters in Devices->Network Parameters.
Then I added a Grandstream HandyTone 286 phone.
The NTP server value shows up as the default. But the syslog server is blank.
If someone else also feels this is a b
010, at 1:12 PM, Pizza Napoletana wrote:
> I don't know what the CallPilot UI is. So I assume I am using the standard UI.
> (I'll mail you a snapshot soon.)
>
> I also have a somewhat unrelated voicemail question:
> If a user is deleted, shouldn't
> /var/sipxdat
Using 4.1.6-018058 on centos5 ...
On the user's web GUI, when you navigate to "My Information" tab, the IMAP
password is displayed in clear text. It would feel a bit safer, especially if
someone is standing right behind you, to display the password as dots or
asterisks.
Thanks
gt; -Original Message-
>> From: sipx-users-boun...@list.sipfoundry.org
>> [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of
>> Pizza Napoletana
>> Sent: Thursday, March 04, 2010 3:49 PM
>> To: Sipx-users list
>> Subject: [sipx-users] IVR hangup af
Using 4.1.6-018058 on centos5 ...
After a user records and accepts their name or personal greeting by dialing the
voicemail extension, the call gets disconnected immediately. Is this the
expected behavior?
I was hoping that the IVR lady would say something nice like "Your name /
personal gree
conference room phones scattered around
the building to sipx. I just noticed that sipX can auto-provision Grandstream
HandyTone 286. It costs $30. I am going to give it a shot.
On Mar 2, 2010, at 5:59 AM, Scott Lawrence wrote:
> Pizza Napoletana wrote:
>
>> In working with Patton s
In working with Patton support on debugging an issue with their M-ATA device (1
port FXS) where it ignores INVITEs from sipx, we have a theory. We think that
the device is getting confused by sipx's TCP based iNVITEs which precede the
UDP based INVITEs.
How can I prevent sipx from using TCP as
Thanks for your help.
I have tried versions "SIP version 4.01.001 OE EN MA (0309)" and a later one
that I can't get access to at this moment.
The support engineer didn't walk me through the debug process, although I see a
setting that allows me to set a debug server.
The device is a single port
Has anyone had success using Patton SmartLink M-ATA device with sipx? I can
register and make calls from the device. But when I call it from another
extension, the device fails to respond to INVITE. It just keeps quiet and sends
no response to INVITE.
I have been in touch with Patton support to
ssage-
> From: sipx-users-boun...@list.sipfoundry.org
> [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Pizza
> Napoletana
> Sent: Monday, February 22, 2010 4:53 PM
> To: Tony Graziano
> Cc: sipx-users@list.sipfoundry.org
> Subject: Re: [sipx-users] 4.1.6.018058
Thanks.
Yes, I did an in-place upgrade on another machine last night and that went OK.
The ISO install on another test machine is where I am having this issue.
I should probably install the old version and do an in-place upgrade to get
around this.
btw, if anyone wants to look into it...
The in-
Yes, a couple of times.
Thanks
On Feb 22, 2010, at 1:22 PM, Tony Graziano wrote:
> Can one assume you rebooted after the install?
>
> On Mon, Feb 22, 2010 at 4:00 PM, Pizza Napoletana wrote:
> Has anyone successfully run the 4.1.6.018058 ISO?
>
> After install, when I visit
Has anyone successfully run the 4.1.6.018058 ISO?
After install, when I visit the web GUI, I get:
HTTP ERROR: 404
/sipxconfig Not Found
RequestURI=/sipxconfig
Powered by Jetty://
sipxconfig.log says the following:
"2010-02-22T20:51:11.619000Z":1:JAVA:INFO:pbx.phone.xxx.yyy.com:main::D
I would like to test a couple of fixes that were made last week in response to
some issues I reported. But the main build at
http://www.sipxecssw.org/temp/sipXecs/ is from Feb 11.
Wiki page at http://www.sipfoundry.org/downloads.html says that dev build is
released every night. Am I looking fo
And, it wiped out the user's Unified Messaging email address setting too!
On Feb 18, 2010, at 5:44 PM, Pizza Napoletana wrote:
> Testing 4.1.6 IM feature.
> I enabled instant messaging for a user from Users->->Instant
> Messaging menu option in the GUI.
> After that, I
Testing 4.1.6
I set a Primary and an Additional email address to be notified in Users->user
id->Unified Messaging screen.
The mails go out fine to the two separate email addresses. But if you check
sipxivr.log, it reports that both emails were sent to the primary address. The
log message is ina
Testing 4.1.6 IM feature.
I enabled instant messaging for a user from Users->->Instant Messaging
menu option in the GUI.
After that, I found that this action changed the first name and the last name
of that user to "unknown".
___
sipx-users mailing li
I have the IMAP sync working correctly in my preliminary tests (except when
deleting via the web GUI), and the only messages related to IMAP that I see in
the sipxivr.log file are the ones you mentioned - i.e., the smtp message being
sent with imap formatting, as follows:
sipxivr.log:"2010-02-1
e bug or the other every day. Must be bleeding edge!
Thanks again for your helpful suggestions.
On Feb 17, 2010, at 1:13 PM, Scott Lawrence wrote:
> 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 NOT
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
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
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
On Feb 14, 2010, at 4:05 PM, M. Ranganathan wrote:
>> The cSeq number of the BYE is wrong. In the trace you posted it is
>> 658568416
>> The INVITE is 1
>> The BYE must have a CSeq number of 2. Hence it is invalid and has been
>> rejected.
[...]
> Perhaps you can report the problem to the ITSP.
R
On Feb 10, 2010, at 8:34 PM, M. Ranganathan wrote:
> I looked at your trace finally. It would be worth trying out 4.1.6. My
> tests indicate it should work.
> Ranga
Hi Ranga,
I tried 4.1.6. I still don't see sipxbridge sending a CANCEL after a successful
REFER. I have attached the log for your re
Please ignore my previous mail regarding Install Updates. There is no problem
there other than myself.
On Feb 14, 2010, at 7:20 AM, Pizza Napoletana wrote:
> btw, no matter how many times I ask it to Install Updates, I keep getting
> "New software updates available" message wi
rlib4.0.4-017289
4.1.6-017914
<> java-1.6.0-sun 1.6.0.7-1jpp
1.6.0.14-1jpp
On Feb 14, 2010, at 7:11 AM, Pizza Napoletana wrote:
> I had 4.0.4 stable ISO running fine on a test machine. I yum'd th
I had 4.0.4 stable ISO running fine on a test machine. I yum'd the latest main
and the machine is now on 4.1.6-017914.
I'd like to test some things on the main build. But some services aren't
starting right. Any suggestions on what I can do to get rid of these issues?
Here are the messages from
All our sipx users are configured to have their own DID alias. When an outside
call comes in and the user doesn't answer, it goes to the user's voicemail. If
the caller really needs to speak to a human, they press 0 and it sends them to
the operator. All this is good.
But when that operator do
On Feb 9, 2010, at 9:32 AM, Dale Worley wrote:
[...]
> Looking at the traces, I see that 5907 receives a REFER to 5903, and
> then generates an INVITE to 5903. What I don't see is any attempt by
> 5907 to CANCEL that INVITE after (as you say) 5907 is hung up. So it
> looks like the originating pho
I am sorry for being pushy and not giving up after sending two messages that
got no response. It may be due to bugs outside of sipx. But any pointer will
help as I have been struggling with this for a couple of days. Here is the
problem again:
> Step 1: Internal user 5907 calls 85907 and gets t
My test environment is sipxecs 4.0.4 from the ISO distribution, connected to an
ITSP via sipxbridge and a bunch of snom 870 phones at the extensions.
I ran into an issue today and I wonder if someone may have ideas on how I can
solve it. Here is the situation:
A call comes from the ITSP and it
http://sipx-wiki.calivia.com/index.php/HowTo_configure_Snom_M3_Wireless_SIP-DECT_phones_with_sipX
On Feb 4, 2010, at 2:59 PM, Johannes Vanderknyff wrote:
> I have one connected to Asterisk. No problems overall, has speakerphone built
> in. It is surprisingly light.
>
> Johannes
>
> On Thu, Fe
If a call comes in from an ITSP and sipX forwards it back to another outside
number via the same ITSP, will this call:
A. Consume two trunks, or
B. Consume zero trunk because the ITSP can automagically hook up the
original caller to the final destination and avoid the loop to our
On Feb 1, 2010, at 6:57 AM, Scott Lawrence wrote:
> On Mon, 2010-02-01 at 09:49 -0500, Jeff Gilmore wrote:
>> This works, but has the minor problem that sipviewer does not accept a
>> filename parameter, so I have to manually navigate to and open the
>> trace file.
>
> It should accept a file name
When a call comes in from an ITSP, via sipXbridge, who sets the Max-Forwards
header in the SIP messages? Does sipXbridge merely decrement what is originally
received from the ITSP, or does it set a new number on its own?
Per logs, sipXbridge's first INVITE has Max-Forwards=8. When forwarding is
the SIP trunk to be used in the case that an
>> internal user calling out doesn't have one assigned
>>
>> is it really that they don't support this or am I doing something wrong ?
>>
>> -gabriel
>
>
> You can try leaving P-Asserted-Identity blank
Is there an easy flag to turn on that would make all the /var/log/sipxpbx
messages go to a syslog server, instead of writing local files?
I'd like to have all the logs from systems under my watch in one syslog server.
Maybe I am missing something obvious with the settings?
btw, regarding that th
On Jan 24, 2010, at 6:52 PM, M. Ranganathan wrote:
>> But Speakeasy gave me a whole bunch of parameters when they provisioned the
>> trunks. Here is what they gave (which I think is for asterisk):
>> ...
>> insecure=very
> No idea what "insecure=very" means but that does sound frightening. :-)
This is my first time configuring sipx with an ITSP. I am sorry it is a newbie
question, but I don't see Speakeasy in the tested ITSP list. I'd like to ensure
that I configure and test it right (along the lines of a test case spreadsheet
I found on the sipx wiki) before I send my config and resu
Has anyone here had success using Siemens Gigaset A580IP DECT base and phones
with sipX 4.0.4? I have a base with 3 handsets, each registering as a different
sipX user. They register fine. But, I have played with them almost all day
today and getting very inconsistent results when making calls.
We are on the verge of deploying sipXecs for two sites. We have a bit of wiggle
room on the final deployment date. If 4.2 is around the corner, I'd rather wait
and deploy 4.2. Does anyone have a sense of when we might see 4.2? There are
some features in 4.2 that will be very useful for us.
I ch
I don't see Speakeasy in sipxbridge's list of interoperable ITSPs. If anyone
has tried sipXecs with Speakeasy's SIP trunks, will you please share your
experience?
Thanks
___
sipx-users mailing list sipx-users@list.sipfoundry.org
List Archive: http://lis
I am a newbie to sipXecs and everything VOIP. I'd appreciate any help I can get.
I just installed the 4.0.4 ISO. But, sipXecs can't discover the spanking new
Snom 870s that I have on the network. Reading some posts from back in May 2008,
I gather these phones may not be supported yet. Is there a
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