Hi Thorben,
Thorben Jensen wrote:
Hi Kong,
No, I have no support for monitoring of Zap devices at the moment. If there
is great demand for it, I will make it.
I would also like some zap monitoring as well. Does it do IAX as well?
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Ken D'Ambrosio wrote:
I'm RTFM'ing, but I can't figure out how the dhcpd.conf file specifies the
boot server, and how it differentiates between whether it's FTP or
TFTP. I've
tried option 66/next-server, and option 150, to no avail. And the docs
just don't -- leastwise, in the way I'm reading
Kevin P. Fleming wrote:
Eric Wieling wrote:
This was fixed in 1.4.1. TFTP and FTP now work the same for deciding
to download the firmware or not.
Interesting... I'll stick with FTP anyway, since I can partially secure
it, and it works across NAT :-)
Hi,
Yeah, I'm still using FTP, it seemed
Hi Androtech,
Androtech wrote:
Dear All,
I bought one IP PHONE from Integrated Networks which was showed to wiki
too:
http://www.voip-info.org/tiki-index.php?page=Intergrated-Networks
I have problems with the Asterisk authentication. It does't want to LOG
IN to Asterisk; it always says LOG ON
Kanuri, Seshu (Company IT) wrote:
Marshall,
I am interested in seeing what you wrote to manage MySQL database
objects.
By the way, latest version of OpenOffice comes with a MySQL
Administrator GUI to manage tables and data. This is something to look
at too.
Seshu Kanuri
I am also interested in
Greetings,
I have a question regarding setting the CallerID, more specifically the
Caller Name. In all of my menus I set the current Caller Name so it
displays what menu they are in when the phone rings for my users. We run
seperate companies so it's easy for us to distinguish how to answer the
misleading warning
messages. It was discussed on the CVS list a month or so ago.
Matt
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http
set
echo - file size: `$ls $destf5`
echo
else
echo no screen logfile to rotate
echo no screen log to give permissions to
echo
fi
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VOIP Administrator
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Justin Carlson wrote:
what was wrong with logrotate?
nothing, i just like doing things my own way :)
this makes use of the asterisk rotate feature, and my own daily log
rotating. meh. to each their own :)
matt
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out if this is some polycom limitation, bug, or my error.
Thanks,
Matt
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http
subscribecontext=context_name
and in extensions.conf add a hint (info can be found near the bottom
of the following page)
http://www.voip-info.org/wiki-Asterisk+standard+extensions
Hope this helps!
Matt
--
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VOIP Administrator
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.
Can you help me to compile such a list?
hi, maybe 1-900, 1-976, i'm not sure of others.
Matt
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to the buddy list when they are added to the contact list. The only thing
I could think of that limits the number to 7 is that it corresponds to the
number of lines available for the IP500. It would make sense that you are
only allowed to monitor 7 since the phone can only handle 6 calls max.
--
Matt
. Or try one of
the Asterisk Live cd's, or customized ISO installers.
I'm not scared to compile asterisk, but I'm not at all interested in
recompiling a linux kernel.
You are going to have to get your hands dirty if you wish to accomplish
anything productive. Or, hire a consultant.
Matt
--
Matt
[tos 0xc0]
18:32:20.394393 10.0.1.96.6000 asterisk.servers.spizzo.spazzo.6801:
udp 42
18:32:20.394549 asterisk.servers.spizzo.spazzo 10.0.1.96: icmp:
asterisk.servers.spizzo.spazzo udp port 6801 unreachable [tos 0xc0]
Matt
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1.314.480.4550 ex
})
exten = s,6,Hangup
exten = s,103,Voicemail(b${ARG1})
exten = s,104,Hangup
Thanks,
Matt
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FWD: 472645
IAXTEL: 1.700.761.1828
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.
This is working for me already... on my polycom IP300 on my desk...
I recently used the asterisk-addons to add mysql to my asterisk, it
looks up CND to find the client name, and then displays both data.
Could you show us an example of this?
Thanks,
Matt
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PSTN
Henry Devito wrote:
BTW they also an iax2 ATA
Try here for the iax2 phone
http://www.ngtel.de/products.php#1
Do you have a contact email for these guys? I couldn't see anything
listed on their site anywhere. Seems the site is in current development.
Matt
Henry Devito wrote:
Hi Matt,
I was just getting ready to try to order a IP phone and ATA in the
morning. This is the contact info I have.
a.. email: [EMAIL PROTECTED]
a.. Phone: +49 69 949 44 185
a.. Fax: +49 69 949 44 118
Thanks for the info, I also saw www.iaxtalk.com is advertising on -biz
Michiel van Baak wrote:
On 05:14, Tue 08 Feb 05, Mazhar Hussain wrote:
If this sounds usefull to you, reply so on the list and I
will try to setup a clear txt doc where and how to find the
sourcecode.
I would like to see the information you can provide on this.
Thanks,
Matt
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commands? Or somehow hijack the SIP session coming
from the server? Just trying to understand the security implications of
allowing the phones to be rebooted remotely (which is a big plus imho).
tia
Matt
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this isn't really asterisk related but it's just damn weird,
anyone have any ideas before I beg radioshack for a return? I've only
had the phone for about 3 months so don't see how it could die already.
TIA,
Matt
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PSTN: 1.877.999.4678 ex. 6400
FWD
Hi Andrew,
Andrew Kohlsmith wrote:
On February 14, 2005 01:18 am, Matt Gibson wrote:
It can receive calls both when receiving power, and when not receiving
power. It can make calls only when not receiving power from the wall. I
tried unplugging it for a good 10-15 minutes to make sure
it was off
Greetings,
I have a server I'm working on here with two tdm cards in it.
4 FXS and 4FX0. Both cards work fine on their own. The problem
lies with using both in the system at once. I have verified the
IRQ's are fine. I have tried switching the slots the cards reside in, no
luck though. I am using
Hi Greg,
Cirelle Internet Products wrote:
In zconfig.h (in the zaptel directory) there is a line at the bottom (at
least in my cvs version)
#define TDM_REVH_MATCHALL
I tried this, then followed with make clean ; make linux26 ; make
install reboot, but it did not function. Then I decided to
Still not working -
I did notice something kinda weird tho, After adding
{ 0xe159, 0x0001, 0xa900, PCI_ANY_ID, 0, 0, (unsigned long) wctdmh },
to wctdm.c, and rebooting
when I issue lspci -v, the PCI id on the card has changed (?). Is this a
normal thing to happen?
Instead of being 0xa900 it's
Hi Nathan,
Nathan C. Smith wrote:
I'm running asterisk stable 1.0.5 and I'm trying to get the netweb eezee
phone version v1.37.008 to talk IAX to asterisk. The pages I saw in the
Try the wiki, myself and someone else wrote up a pretty big howto and
tips and tricks on these phones.
information, it'd be much
abliged.
ps: i'm using slackware 10.0, and kernel 2.6.9, and a tdm30b, and a
x100p clone.
thanks,
Matt
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talked with Sayson, and they seem to have 'forgotten' about me... I
bought my phone from Radioshack here in Canada.
Could you send me the codes you guys have working so I can test them
with my phone?
Currently I have:
SECURITY 0x0106
FDN 0x0106
for slot 4.
Matt
--
Matt
in the same context?
matt
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' the phone and then the
issue never arose again. Dunno if that's what you need to do, but it
helped me :)
Hope you get it working,
Matt
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itself and was fixed.
probably not the right way, but definately quick n easy :)
matt
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http
any of the other phones. All of the other phones are IP500's with
the newest public firmware release.
Thanks,
Matt
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[EMAIL
Chris W wrote:
Any suggestions welcome.
Do you have the proper FDN/SEC codes for your phone located in
asterisk.adsi, and have an extension created to program your phone?
Matt
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-Realtime in it's current state?
Matt
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$asteriskopts
Matt
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Recently there was talk about NANPA and getting current info from them -
I found this link and thought I should share..
http://bellsmind.net/NANPA/
Matt
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I have a script that I wrote that actually stops and restarts the server
(asterisk -rx stop now su asterisk -c /usr/sbin/safe_asterisk).
It works fine on slackware, but not on fedora, not too sure why. You're
welcome to use this if you want.
script
#!/bin/bash
# must be running asterisk
in extensions.conf (it didn't seem to work)
exten = 9056742007,1,Answer
exten = 9056742007,2,Wait,1
exten = 9056742007,4,Queue(tech|hH)
exten = 9056742007,5,Voicemail(u9056742007)
exten = 9056742007,6,Hangup
Thanks
Bill
___
--
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Hi James,
I have one posting for the Cisco7970 ringtone, which you can adapt for
the Polycom. It's here: http://www.voipphreak.ca/archives/349
I also have another one I posted for the Polycom Ringtones with a
bunch of tunes. It's here:
http://www.voipphreak.ca/archives/78
Hope these help :)
Maybe I'm wrong, but don't you have to stop/start asterisk for
voicemail changes to take effect on 1.2 (like zapata)
Matt
On 12/10/2007, Jesse Scott [EMAIL PROTECTED] wrote:
Doesn't look like FreePBX is nuking it. I just SSH'd in and opened the
voicemail.conf directly and the entry is still
How come he has it, and he's in Paris! I'm in Toronto, and I don't have
it?
:(
I was thinking the same thing, Ottawa here.. :(
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Not sure if it's related, but I experienced similar problems with Utorrent
WebUI.
http://developer.mozilla.org/en/docs/DO
http://developer.mozilla.org/en/docs/DOM:stylesheet.href . sheet.href
Mozilla updated their spec for inline styles. Unsure if this is the cause of
your issue, but it
Hi Remco,
Both of these may be helpful to you, one to fix the SMS issue, and one to
enable the stock MS voip client:
http://www.mattgibson.ca/2008/04/13/fix-sms-time-issues-on-rogersfido-unlock
ed-gsm3g-windows-mobile-56-smartphonespocketpcs/
and
Hi Roland,
Did you try:
http://www.voipphreak.ca/2008/03/29/enable-the-hidden-voip-features-of-windo
ws-mobile-6x-for-free-voip-calls-using-asterisk/
We have this successfully working on a Touch (ELF), and a HTC Tilt (Tytn II)
Thanks,
Matt G
: http://www.voipphreak.ca
:
Hi Roland,
No problem, glad this works for you. We don't find it too bad.
Hm, I'm not sure why you're having difficulty with the editing tool, you can
check on xda-developers.org forum for more information on the editing tool,
there may be a newer version. If you need help, feel free to
] Windows Mobile 6 IAX/SIP client?
On Thu, 3 Jul 2008, Matt Gibson wrote:
http://www.voipphreak.ca/2008/03/29/enable-the-hidden-voip-features-of-windo
ws-mobile-6x-for-free-voip-calls-using-asterisk/
Thanks for the link!
I installed and configured the phone according to the above link.
It only seems
voicemail.
When i try to call any other number i can see that the phone is dialling a
9 before the number i want to dial.
Weird..
On Sun, 6 Jul 2008, Matt Gibson wrote:
Hi Remco,
Here's my SIP config..
[8902]
type=friend
secret=xxx
record_out=Adhoc
record_in=Adhoc
qualify
Hi Remco,
or maybe i am using an old voipwm6.cab or sip config? I also seem to have
the problem that sound is only coming from the speaker on the back, not
the ear
speaker.
Which specific phone do you have? Your best bet is to probably check:
http://forum.xda-developers.com/
Hi All,
Apologies for this, migrated the site and forgot to change a path. Site's
back up now.
Thanks,
Matt G
: http://www.voipphreak.ca
: http://www.ratemydialplan.com
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Paul Hales
Sent: Sunday, July 27,
I've used http://www.pbxprompts.com/
The whole pack is around 100$ and then I think I was charged 11$ per prompt
for custom ones. No setup fee that I recall.
Thanks,
Matt G
: http://www.voipphreak.ca
: http://www.ratemydialplan.com
From: [EMAIL PROTECTED]
[mailto:[EMAIL
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Lyle Giese
Sent: Tuesday, August 05, 2008 6:43 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] email notification to external email address
Brian Simpson wrote:
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Patrick
Sent: Wednesday, August 06, 2008 7:34 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] OT: Cisco 7961 SIP downgrade from 8.3.3 -
8.0.4SRS2 failing
Hi,
My apologies for the OT. My
Hi JR,
This may help you - we were using it to route calls from friends through the
IVR so they hit us directly. You'll have to modify it to suit your dialplan,
but it should be a good starting point.
http://www.voipphreak.ca/2006/11/26/asterisk-14-php-rolodex-howto-script/
Thanks,
Matt G
:
Let me know if you find out - We played around with this for a while but
could never get it to work. We ended up sending multiple messages with blank
lines to get the spacing we wanted.
Thanks,
Matt G
: http://www.voipphreak.ca
: http://www.ratemydialplan.com
-Original Message-
From:
I noticed one thing,
/etc/dahdi/system.conf:
loadzone = us
defaultzone=us
fxoks=1,2
fxsks=4
echocancel?
Tzafrir mentioned it earlier, but it may have gotten lost on the thread. I
was having problems with Dahdi until I added echocancel to our system.conf,
could this be your problem?
As
It seems to me the dahdi driver works. For some reason, however,
chan_dahdi doesn't see the channels the driver set up.
Anybody else using TDM400P with dahdi and rc4?
Hi Sean,
Not sure if it matters, but we're using 2.0R3, noticed you're on 2.0R2
Unfortunately, I don't have an actual
Did you setup the new /etc/dadhi/system.conf as well as unloading your old
zaptel modules and re-inserting the new dahdi modules?
* The primary kernel modules have changed names; the new names are:
zaptel.ko-dahdi.ko
ztd-eth.ko - dahdi_dynamic_eth.ko
-Original Message-
From: John covici [mailto:[EMAIL PROTECTED]
Sent: Thursday, September 11, 2008 1:52 PM
To: Matt Gibson
Cc: [EMAIL PROTECTED]; 'Asterisk Users Mailing List - Non-Commercial
Discussion'
Subject: RE: [asterisk-users] dahdi vs zap with latest version of asterisk
-- having
We have done something similar using the category option with the voicemail.
Our emails look like this:
--
TO : Big Boss
ID : 2
CAT. : EMERGENCY
BOX : 100
FROM : Emergency Line 5552221212
DUR : 0:20
DATE : Wednesday, October 10, 2007 at 01:28:27 PM
--
Internal
From the doc/sip-retransmit.txt
What is the problem with SIP retransmits?
-
Sometimes you get messages in the console like these:
- retrans_pkt: Hanging up call XX77yy - no reply to our critical packet.
- retrans_pkt: Cancelling retransmit of
Do you have ztdummy loaded in the VM?
Thanks,
Matt G
: http://www.voipphreak.ca
: http://www.ratemydialplan.com
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michael J.
Liberatore
Sent: Wednesday, September 24, 2008 8:28 PM
To: Asterisk Users Mailing List -
Check our howto:
http://www.voipphreak.ca/2007/04/16/monitoring-asterisk-14-with-snmp-and-cac
ti-for-pretty-graphs/
and for nagios monitoring
http://www.voipphreak.ca/2008/06/19/monitoring-asterisk-with-snmp-nagios-and
-nagios-administrator-using-ubuntu-lts-804-server/
Thanks,
Matt G
:
This may help:
http://www.voipphreak.ca/2008/03/29/enable-the-hidden-voip-features-of-windo
ws-mobile-6x-for-free-voip-calls-using-asterisk/
Note, that most sip clients for WINMOB suck and send the voice out the back
speaker instead of the front speaker. I've found one other client (can't
.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt Gibson
Sent: October 3, 2008 11:52 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] sip clients for smart phones?
This may help:
http://www.voipphreak.ca/2008
on TouchFlo3D. I don't see where to even use
it.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt Gibson
Sent: October 3, 2008 3:34 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] sip clients for smart phones?
I
This may be what you're looking for:
http://www.linuxjournal.com/content/custom-checks-and-notifications-nagios
Thanks,
Matt G
: http://www.voipphreak.ca
: http://www.ratemydialplan.com
: http://www.asterisk-jobs.com
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
Did the server reboot or lose communication? This happens with our 7970's
sometimes if there's been a hiccup, usually dialing voicemail registers them
back up - occasionally we've had to do the soft reboot from the screen.
401 unauth - looks like it may be md5secret issue, or nat traversal over
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jerry Geis
Sent: Tuesday, October 07, 2008 5:42 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] cisco phones getting SIP 401 unauthorized
Matt,
The phones are inside the LAN.
what is
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jerry Geis
Sent: Wednesday, October 08, 2008 10:13 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] cisco phones getting SIP 401 unauthorized
Hi Jerry,
Hm, okay. We had to use
Are you sending SMS to known users or to any mobile phone user?
If you are sending to a fixed user base, track down the email to SMS
gateways for their carriers. Then sending an SMS is no different than
sending an e-mail.
If it's for something really important this might not be the
after a fresh installation of Freepbx
1- How can i make Freepbx send voicemail to Email. (the Linux mail
configuration steps)
2- How can i operate Fax machine and How it will be able to send the FAX to
email.
3- Is there any software application i can run to monitor live the calls and
to see
Hamilton
Sent: Friday, October 24, 2008 8:57 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Fresh installed box
queuestats?
Original Message
Subject: Re: [asterisk-users] Fresh installed box
From: Matt Gibson [EMAIL PROTECTED
: [asterisk-users] Fresh installed box
queuestats?
Original Message
Subject: Re: [asterisk-users] Fresh installed box
From: Matt Gibson [EMAIL PROTECTED]
Date: Fri, October 24, 2008 6:16 pm
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users
Try using SSMTP
http://www.linux.com/articles/132006
It works with any provider for mail sending, and takes 30 seconds to setup.
Thanks,
Matt G
: http://www.voipphreak.ca
: http://www.ratemydialplan.com
: http://www.asterisk-jobs.com
-Original Message-
From: [EMAIL PROTECTED]
I am also working on this, and have a marketing/communications
background. I may be able to help cheaper than the big agency :)
thanks,
matt
On 20/04/07, dave cantera [EMAIL PROTECTED] wrote:
robert,
I might be interested depending on cost, message, and quality...
keep me in the loop.
daveC
Hi All,
As the subject describes, has anyone gotten this to work? I am running
an asterisk 1.2.16 server, and am trying to register my cisco 7970
remotely to it, but it just won't go.
I am running 1.4.2 internally and the phone registers fine to it. I'm
using the latest firmware (i think) -
: 3600
Contact: sip:[EMAIL PROTECTED]:5060;transport=udp;expires=3600
Date: Tue, 24 Apr 2007 21:40:09 GMT
Content-Length: 0
Thanks for your help!
On 24/04/07, Matt Gibson [EMAIL PROTECTED] wrote:
Hi All,
As the subject describes, has anyone gotten this to work? I am running
an asterisk 1.2.16
generated the password with echo -n 125:asterisk:pass | md5sum
Thanks,
MG
On 24/04/07, Matt Gibson [EMAIL PROTECTED] wrote:
Here is a followup:
I've now tried SIP 7.0.5 which also doesn't work. I've also got
debugging information from both sites (1.4.2, nat, local) and (1.2.16,
no nat, remote
Hi Everton,
Which portion of my howto are you having trouble with? Make sure you
have compiled the res_snmp when you compiled asterisk. If you want to
take this offlist email me at [EMAIL PROTECTED]
Thanks,
Matt G
On 02/05/07, Everton Goularth [EMAIL PROTECTED] wrote:
Hi,
I`m trying to use
Hi Sanjay,
This is easily fixed.
Check this bug report for how to fix it:
http://bugs.digium.com/view.php?id=8575
Thanks,
MG
On 13/04/07, Sanjay Rajdev [EMAIL PROTECTED] wrote:
I have a CISCO 7912 phone, the LED on the phone does not glow when there is new
voicemail, can we configure
Which grandstream phone should I buy, this is going to be for small
office for testing purposes.
I am on a budget, hoping to find someone here who has some used to
sell or point me in the direction of a seller.
Hi Mike,
If you're set on the Grandstreams, and it's just for testing the
We're using it here on dynamic IP from our ISP.
They provide reverse DNS, which we've simply setup a CNAME to.
So, CPE390480Q239432098423.MYISP.COM is cnamed to PBX.MYBUSINESSDOMAIN.COM
Did not have to change anything else for this to work.
Thanks,
Matt G
: http://www.voipphreak.ca
:
In my experience cepstral has always had much nicer sounding voices, but I
haven't tinkered too much with either. There is a reason one is pay and one
free though J I believe cepstral is still offering demo's, I'd download each
and see which one gives you the performance you're looking for.
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of OCG Technical
Support
Sent: Wednesday, December 03, 2008 11:14 PM
To: 'Asterisk Users List'
Subject: Re: [asterisk-users] Windows Mobile 6 SIP client: Remote hostcan't
match request NOTIFY to call
You’ll have to recheck your
We often find ourselves reading through all sorts of contests on the
Internet that never seem to echo our own personal skill set or interests.
Perhaps you've even fantasized about a type of contest with the types of
prizes and goodies that YOU'D actually enjoy. Maybe you've wished there were
selected as
the winners and will be awarded the following prizes:
[snip]
I think you'd get just as much interest in an Obfuscated Dialplan
Contest which seems to be the most popular type of dialplan
programming. The more unreadable, ugly, and opaque the code becomes,
the more
3rd place: An APSTel dial plan (standard license) donated by APSTel!
So... if you can write the slickest dialplan, you get dialplan
generator
software?
Hi Andrew,
Well, the thought is that most people are using SmartDraw, Dia, Visio,
Illustrator or Corel Draw to create these types of
http://www.voipphreak.ca/2007/04/16/monitoring-asterisk-14-with-snmp-and-cac
ti-for-pretty-graphs/
Thanks,
Matt G
: http://www.voipphreak.ca http://www.voipphreak.ca
: http://www.ratemydialplan.com http://www.ratemydialplan.com
: http://www.asterisk-jobs.com
# cd /directory
# rm -rf *
Thanks,
Matt G
: http://www.voipphreak.ca
: http://www.ratemydialplan.com
: http://www.asterisk-jobs.com
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David @ULC
Sent: Monday, February 23,
Have you tried using md5secret, not sure if that will do it - but that's how
we had to get our 7970 registered with freepbx/trixbox - unfortunately they
don' t have this ability built in (yet). I have a patch if you need it,
contact me off list. As a quick test you could enable it in the config
Give www.asterisk-jobs.com a try too if you want J it's free.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Sean McMaster
Sent: Monday, March 09, 2009 3:47 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Job in
qualified individual.
Please keep the comments coming, we enjoy hearing what
users think good or bad, either way in the end it helps the
site to be a better place for people to find the employment
they want.
Thanks,
Matt Gibson
Asterisk Jobs
On 22/08/06, Peter Bowyer [EMAIL PROTECTED] wrote
Hi Guys,
I thought some of you might be interested in a minimalistic Polycom
ringtones howto.
I assume this works with the ip600 (501/601) but not sure about the 300.
http://www.voipphreak.ca/archives/82-My-Little-Howto-for-Polycom-IP500-Ringtones.html
Matt
--
Matt Gibson
Telecommunications
tim panton wrote:
A Physicist friend of mine named all the machines in a class C after
the chemical elements (Hydrogen was x.x.x.1 etc).
Our work systems are named after the (fictional) islands in the Earthsea
books.
That's an interesting idea.
Currently most of my servers are either 'word
Hi Matt,
Matt Darnell wrote:
From the Wiki:
'There are aready a few bugs in 1.5.2 but more fixes some good new
features'
Anyone know where the bugs are being listed?
I am working through a few issues:
1. When rebooting, the phone will pause for exactly 180 seconds with
the screen reading
Chris Coulthurst wrote:
Does anyone know how to change the display format of caller id on the
screen of a polycom 300/500/600?
Hi Chris,
Edit your MAC-directory.xml for each phone
and instead of using the fn/fn and ln/ln
only use fn
(this equates to only using the first name and not the last
Hi Nathan,
Nathan E. Pralle wrote:
Greetings all.
Well, the first Asterisk Sound List in HTML was so popular, I did some more
fiddling around to make it even more useful. Here's an updated page:
- One master list with all sounds, sorted alphabetically by filename
- The old lists are linked
Michael George wrote:
On Sat, May 28, 2005 at 11:10:30AM -0400, Steve Totaro wrote:
qualify = yes is what is causing the messages. You can assign a value
rather than yes. like 1000 or something or you can remove the qualify
statement alltogether. The message is just a warning. Eliminating
1 - 100 of 139 matches
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