OK close to last mail before vacation ;)
We had some issues with wav to g722 conversion in freeswitch before the early
commits after 1.0.6. Iw ill send you sipx-freeswitch rpms when I get back next
week to test if you like. Just drop me an email which centos version etc you
are using o if its
Masochist :P
I think the whole idea of doing a freeswitch controlling voicemail in java is a
bit of an overkill. My suggestion was to replace this by a far more
lightweight lua script within freswitch, which would give the options of rapid
changes and fixes and add there multi file format
Il martedì 10 agosto 2010 20:48:41 Tony Graziano ha scritto:
according to snom they only support
G.711, G.729A,
G.723.1, G.722, G.726,
GSM 6.10 (full rate)
on that device. your itsp is making offers not even in that list. I'm
guessing that is an issue but I still need a useable siptrace to
Michal
I am sure you looked into the way we ended up doing voicemail. The decision
to not use VM in FS but create a new one at the time was based on the need
to have compatibility with what sipXecs provided previously. Therefore, the
first step was to replicate the old system on FS.
On Wed, Aug 11, 2010 at 7:32 AM, Martin Steinmann mstei...@gmail.com wrote:
I am sure you looked into the way we ended up doing voicemail. The decision
to not use VM in FS but create a new one at the time was based on the need
to have compatibility with what sipXecs provided previously.
I wrote the original wiki page on the m3, which snom oem's from another
manufacturer with their own version of firmware.
I have not experienced this issue with my m3's, ever.
I assume the m9 is not that much different from the m3.
Please refer to the wiki page:
Thanks, i checked this page, but i will check it again to see if i
missed something.
In status page of the phone web interface i see this, it looks that this
is the firmware version:
VersionSIP=9.1.70 DCM=GLOBALNETFTCL14N20100611,JUL 01 2009
On Wed, 2010-08-11 at 08:10 -0400, Tony Graziano
There is an even newer one. Such is the case with snom, whether good or
bad...
http://wiki.snom.com/Snom_m9/Documentation/Online_Manual#Firmware_Update
On Wed, Aug 11, 2010 at 8:30 AM, Ali Nebi an...@iguanait.com wrote:
Thanks, i checked this page, but i will check it again to see if i
missed
I access the sysadmin portal practically everyday using my Android EVO. It
works great and I haven't come across anything yet that I could not do from
the handheld device.
On Mon, Aug 9, 2010 at 8:12 AM, Sven Evensen sven.even...@onrelay.comwrote:
I thought it would be pretty cool if end
Mike, which browser are you using? And any special settings?
Sven
From: Mike Haun [mailto:haun.m...@gmail.com]
Sent: 11 August 2010 14:10
To: Sven Evensen
Cc: Douglas Hubler; sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] sipXconfig on mobile
You have to zoom in quite a bit to get the drop down menus to work right.
sipXconfig works on my Android phone no problem.
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Sven Evensen
Sent: Wednesday, August 11, 2010 8:34 AM
To: Mike
I access 4.04 on my Windows Mobile 6.1 device using Opera. All I’ve
done is adjust the browser to fit content to the screen.
From: Josh M. Patten [mailto:jpat...@co.brazos.tx.us]
Sent: Wednesday, August 11, 2010 9:35 AM
To: sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] sipXconfig
the voicemail email is already 'suspect'. VERY small body, and
larg(ish) wave attachment.
Anything (including following the relevant RFC's) will help.
# 1 (discussed already), a From of postmas...@localhost can trigger some
anti-spam defenses when sending voicemail.
(postmaster usually denotes
seems like reasonable issue and worthy of tracker item
On Wed, Aug 11, 2010 at 9:58 AM, Michael Scheidell
michael.scheid...@secnap.com wrote:
the voicemail email is already 'suspect'. VERY small body, and larg(ish)
wave attachment.
Anything (including following the relevant RFC's) will help.
I would say this is certainly Michael's realm of expertise - hopefully no one
has any extensions labeled as: FREE VIAGRA(R)
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Douglas Hubler
Sent: Wednesday, August
Ah, and a description of how to make your host allowed with a SPF record
would be good too... there's some good useable help in the system, but
sometimes more is better.
On Wed, Aug 11, 2010 at 10:24 AM, Nathaniel Watkins
nwatk...@garrettcounty.org wrote:
I would say this is certainly
Well, tried removing the +1 from the gateway and added to the individual
dialplans. Local / LD calls still works, but toll free numbers still
doesn't work. It's time for Voxitas to dig deeper to see if they have
made any recent changes or something in Sipx from the recent updates.
Outgoing
Like I said, it could be a voxitas change. If you sign up for a
voip.msaccount and add a dialplan to send certain calls there, do toll
free calls
work? does outbbound callerid work?
On Wed, Aug 11, 2010 at 11:44 AM, Tran, Ly V. lt...@rrtgi.com wrote:
Well, tried removing the +1 from the gateway
On my centos, I have /etc/hosts.deny set to ALL:ALL
Then, I set in hosts.allow for sshd access from certain IPs only. NO other
entry in hosts.allow. SSH seems to be properly locked down.
Looks like sipX GUI config using http (port 80) and then links to https on port
8443 seems to work from
Bang went the nail with one strike. You make a good hammer!
On Wed, Aug 11, 2010 at 11:55 AM, Todd Hodgen thod...@verizon.net wrote:
You nailed it Tony. Thanks for the troubleshooting tip. The trunk was
staying up during the entire message recording, and pressing # did cause the
Im assuming everyone is getting this email on all of their messages to the
sipx-users group. If so, can we remove this address as a nuisance? Its
been going on for a week.
From: mailer-dae...@bjtu.edu.cn [mailto:mailer-dae...@bjtu.edu.cn]
Sent: Wednesday, August 11, 2010 8:56 AM
To:
i never noticed that. they are actually in my spam folder though.
On Wed, Aug 11, 2010 at 11:58 AM, Todd Hodgen thod...@verizon.net wrote:
I’m assuming everyone is getting this email on all of their messages to
the sipx-users group. If so, can we remove this address as a nuisance?
It’s been
Content-Type: text/plain;
charset=utf-8
Content-Transfer-Encoding: 8bit
Organization: SipXecs Forum
X-FUDforum: 08063afcdd00a6e76393c5b9527381e8 50273
Message-ID: c461.4c62d...@forum.sipfoundry.org
First of all I'm sorry for posting this twice. I forgot to
subscribe to the mailing list.
I'm
Matt, I assumed it was just me that got these. I guess we just remove
this person. Is there any way to divert these types of emails to
folks that have access and know how to remove people from list?
-- Forwarded message --
From: Todd Hodgen thod...@verizon.net
Date: Wed, Aug
Can you post the 420 msg?
This is usually caused by a particular SIP header that it may not like. It
should write the unsupported header in the 420 msg as per RFC 3261.
Like this
Supported: Privacy
Which means the device does not support the Privacy related header.
Is your Phone sending a
According to Voxitas, this is what we have to do to fix this problem:
We need to have the 11 digit number inserted into the From header field
of the call in the initial INVITE. Right now the P-Asserted-Identity is
showing the e164 format which will be perfect for the From header field
also. I
it wont show up until your evo is updated to android 2.2 because thats a
flash thingy
On Wed, Aug 11, 2010 at 2:46 PM, Matt White mwh...@thesummit-grp.comwrote:
haun.m...@gmail.com wrote:
I access the sysadmin portal practically everyday using my Android EVO.
It
works great and I haven't
I figured out:
http://track.sipfoundry.org/browse/XX-7885
Seems to me I am suffering from this?
I will debug sipxrls tomorrow and post a snapshot.
René
2010/8/11 Rene Pankratz rene.pankratz.l...@iant.de
Hello list members,
after updating my PBX and my snom telephones cannot subscribe
Using a HTC Desire (Android 2.2) with the default browser I don't see a play
button when looking for voicemails.
But it seems to me that the layout is not displayed correctly as the text of
duration is cut in the middle. It seems the width of the display is to
small for showing the whole layout.
says it was fixed in 4.16. it also says it affected polycoms, etc. I am
running 4.0.4 and 4.2.0 through 4.2.1 without seeing this (polycom). What
version of sipxecs are you using?
On Wed, Aug 11, 2010 at 3:23 PM, Rene Pankratz
rene.pankratz.l...@iant.dewrote:
I figured out:
get a bigger phone... haha.
i'm really waiting for the motorola version that is being worked on with a
2ghz snapdragon... now if that comes with a foldout led display...
On Wed, Aug 11, 2010 at 3:38 PM, Rene Pankratz
rene.pankratz.l...@iant.dewrote:
Using a HTC Desire (Android 2.2) with the
I have a call coming in via a Sip trunk to an extension assigned to an AA.
AA plays the prompts user to dial 1 or 2...
In either case I send the call back out over the SIP trunk to a Cell PSTN
number. The call connects but I have no Audio either way.
Which Logs should I collect and provide to
What you are describing is a hairpinned call. You should provide a siptrace
of the call with the proxy at debug as a minimum.
You should also describe your environment...
what kind of phone/ua (firmware software version might be relevant), whether
the UA or sipx is behind a nat or if the user is
See inline
From: Tony Graziano [mailto:tgrazi...@myitdepartment.net]
Sent: Wednesday, August 11, 2010 2:37 PM
To: Ujjval Karihaloo
Cc: sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] SIP Trunk -- AA-- SIP trunk call flow
What you are describing is a hairpinned call. You should provide a
Anyone else have issues when calling the default auto attendant (ext. 100) -
Press '9' for the dial by name. The AA voice then states: Please spell the
name of the person (screws up here) press 7 for q... - the audio just garbles
for a second.
(4.2.1-018930 2010-06-04T15:25:27 build34)
On 8/11/10 4:45 PM, Nathaniel Watkins wrote:
name of the person (screws up here) press 7 for q... -- the audio
just garbles for a second.
polycom firmware 3.2.3.
drop back to 3.13c
--
Michael Scheidell, CTO
o: 561-999-5000
d: 561-948-2259
ISN: 1259*1300
*| *SECNAP Network Security
No. Im running what he said. haha.
On Wed, Aug 11, 2010 at 4:56 PM, Michael Scheidell
michael.scheid...@secnap.com wrote:
On 8/11/10 4:45 PM, Nathaniel Watkins wrote:
name of the person (screws up here) press 7 for q…” – the audio just
garbles for a second.
polycom firmware 3.2.3.
drop
I think it might matter what the softphone is, and the version...
I would try the AA using the phantom user just to see if the audio works.
its not a solution, just a troubleshooting step to determine if the proxy
cares or not.
On Wed, Aug 11, 2010 at 4:42 PM, Ujjval Karihaloo
On 8/11/10 7:40 AM, Douglas Hubler wrote:
On Wed, Aug 11, 2010 at 7:32 AM, Martin Steinmannmstei...@gmail.com wrote:
had a user on a iphone report that the voicemail he got using the mp3
format was so much clearer then the wav file.
seems to be working fine now, getting anywhere
okay, so in case there is still someone out there on 3.10.x looking to
upgrade, here is my 3.10.3 - 4.0.4 - 4.2.1 path, which (seems to have)
worked for my single-server setup. 4.0.4 yum repos seem to be borked
(pointing to 4.2.1 repos), which slowed me down a bit.
basically:
(1) remove sipxecs
Software update 4.2.1-18890.6.2 showed up this morning.
I tried to install it with the following results:
http://sipxecs.sipfoundry.org/pub/sipXecs/LatestStable/CentOS/5/i386/RPM/repodata/repomd.xml:
[Errno 14] HTTP Error 404: Not Found
Trying other mirror.
I have the Polycom productivity Suite and am looking at how to enable
RTCP-XR packet generation. We are an enterprise engaged in a Proof of
Concept evaluation of SipXecs, ergo this is not live users - so I can
hack away up to a point, and run what ever version I need.
According to the Polycom
http://sites.google.com/site/sipxecstipsandtricks/polycom-phones
shows you how to patch sipX to allow for custom configuration files
NOTE: wget for some reason is unable to download files from this site so you'll
need to download them from firefox then transfer them from your computer to
your
remove that repo file and get that one you need from here...
http://download.sipfoundry.org/pub/sipXecs/
On Wed, Aug 11, 2010 at 6:56 PM, Norman Branitsky
nor...@cherniaksoftware.com wrote:
Software update 4.2.1-18890.6.2 showed up this morning.
I tried to install it with the following
you need to edit the vm template in sipx to create/default that value to
on.
On Wed, Aug 11, 2010 at 7:21 PM, McIlvin, Don
don.mcil...@nrtnortheast.comwrote:
I have the Polycom productivity Suite and am looking at how to enable
RTCP-XR packet generation. We are an enterprise engaged in a Proof
did you ever get a resolution to this?
On Thu, Jul 29, 2010 at 3:03 PM, Matt White mwh...@thesummit-grp.comwrote:
Attached is a good call trace.
The only thing I see in this good trace is the 200 OK comes back in very
quickly..in about 1ms. In the bad trace the phone goes at least 2 ms
On 8/11/10 9:26 PM, Tony Graziano wrote:
The current unstable build (4.3.0) has (limited) t.30 faxing support
for incoming faces under unified messaging. I will be among several
starting to test this over the next few days. At the same time, FS
1.07 will be introduced into unstable builds in
mp3 guy... I was the original mp3 guy. look how old my tracker item is!
I actually think getting the VM encoded as mp3 might be easier than some of
the twister moves you've done (congrats), and it is now on the roadmap.
votes for that are welcome as well. mp3 might be playing better for you on
an
On 8/11/2010 7:46 PM, Tony Graziano wrote:
remove that repo file and get that one you need from here...
http://download.sipfoundry.org/pub/sipXecs/
That worked correctly, thanks.
Installed:
kernel.i686 0:2.6.18-194.11.1.el5
Updated:
i suppose it will probably have to change again. it was a lot of work for
those guys to get it done this far.
there may end up being a separate directory for testing (unstable) versus
stable, who knows this late at night?
it looks like you did an OS update from your results and not a sipx
On Wed, Aug 11, 2010 at 10:06 PM, Tony Graziano
tgrazi...@myitdepartment.net wrote:
i suppose it will probably have to change again. it was a lot of work for
those guys to get it done this far.
there may end up being a separate directory for testing (unstable) versus
stable, who knows this
On Wed, Aug 11, 2010 at 6:49 PM, milosz mew...@gmail.com wrote:
4.0.4 yum repos seem to be borked
(pointing to 4.2.1 repos), which slowed me down a bit.
We only have repo files for latest stable and testing builds. I
didn't anticipate folks needing repos for upgrading to intermediate
builds
On 8/11/2010 10:06 PM, Tony Graziano wrote:
i suppose it will probably have to change again. it was a lot of work
for those guys to get it done this far.
there may end up being a separate directory for testing (unstable)
versus stable, who knows this late at night?
it looks like you did an
I would think a 4.0.4 repo would be handy. Otherwise one would have to
update a package at a time and hunt down dependencies.
Tony Graziano, Manager
Telephone: 434.984.8430
Fax: 434.984.8431
Email: tgrazi...@myitdepartment.net
LAN/Telephony/Security and Control
On Wed, Aug 11, 2010 at 10:52 AM, Tony Graziano
tgrazi...@myitdepartment.net wrote:
Ah, and a description of how to make your host allowed with a SPF record
would be good too... there's some good useable help in the system, but
sometimes more is better.
SPF always seems like a good idea so
Just covering bases. SPF is a great idea. Getting people to properly deploy
great ideas is another thing.
Tony Graziano, Manager
Telephone: 434.984.8430
Fax: 434.984.8431
Email: tgrazi...@myitdepartment.net
LAN/Telephony/Security and Control Systems Helpdesk:
On 8/11/2010 10:50 PM, Tony Graziano wrote:
Not what I was saying. In the uodate log you sent me I only say OS updates I
think. I was assuming that sipx also updated and was happy it did so.
As listed below, sipximbot did upgrade.
- Original Message -
From: Norman Branitsky
Indeed polycom phones subscribe the list without an error.
But the problem still persists with snom phones.
Maybe I should switch to sipx-dev and discuss if I should re-open the
ticket.
René
2010/8/12 Rene Pankratz rene.pankratz.l...@iant.de
I am using 4.2.1-018930 (clean iso install)
The
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