Hi,
Is it possible to disable the DND feature on a Polycom 501?
Regards,
Jan
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
Hi,
I'm using Asterisk 1.2.1 on a Debian Sarge with a TDM400P and a monoBRI
using chan-mISDN from beronet site.
It seems to work all right except for autodial calls, monoBRI ISDN
channel behaves differently waiting for the caller to answer and then
continue.
Asterisk console says:
analog:
If I call PSTN number a, than I can call the extension number, while
when I call PSTN phone number b the tones are ignored.
If I call PSTN PSTN directly the extension number can be dialed.
How can I improve that?
bye
Ronald Wiplinger
___
I would also recomend that you upgrade to the latest firmware 1.0.2.13
(contact grandstream) as it does fix some registeration issues and have
extra NAT/STUN features.
On Wed, 2006-05-03 at 17:15, Chris Bagnall wrote:
Greetings list,
I'm coming across an issue with some of the GXP-2000 phones
Title: Messaggio
Ihave 13 Snom 320 with asterisk 1.07
bristuffed. The problem is that sometimes on random basis, when one customer is
placed on hold and another call arrives, the customers are put in conference
with each other. This look very strange to me, but I've disabled the confernce
This is a known problem and it does not matter what zaptel timer you use. A
solution is available in 'svn head' by using
asterisk.conf
internal_timing = yes
OR
Enable internal timing support (-I)
on the command line. I don't know if this has been backported to the stable
branch.
Chris
Hi,
I configured a console channel for my sound card and assigned an
extension to it. That way, I am able to talk to any SIP account when
they call this extension. For testing purposes, I now would like to be
able to allow only one specific codec and reject all calls to the
console with
I noticed one thing: I got courtesytone = beep in my features.conf
If I took it off, I got no sound.
That's one sorted out :-)
Do you have this on it? Do you have a global DYNAMIC_FEATURES =
monitor in extensions.conf ?
Yes.
___
--Bandwidth and
On 5/3/06, Dr. Michael J. Chudobiak [EMAIL PROTECTED] wrote:
One of my users reports frequently hearing echo on her Snom 360 phone,
even while talking to other Snom phones (via Asterisk) on the same LAN
(i.e., all-digital low-latency connection). I can never reproduce it
though, and swapping
From this list I found that I could use SetGroup and CheckGroup to do
what I wanted. But I'm not quite sure how I do it.
The case is that I have 3 user groups, and one main group. The main
group is for all the incoming calls from external phones. The main group
should be allowed to have 3 calls
Finally, I decided to turn hyperthreading back on, and everything is back
to normal.
Unless there is somewhere in CentOS 4.3 that has the processor count
hardcoded from the install, I am baffled by this.
Was it on when * and zaptel was compiled?. Maybe the compiler produced HT
optimized
On Thursday 04 May 2006 20:53, Asterisk wrote:
The handsets do not work with the SIP flag to make them AUTO-ANSWER. (As
documented is should)
Ie, you cannot use them with intercom or Page features.
Works fine here;
SIPAddHeader(Call-Info:\;answer-after=0)
hads
--
You buttered your bread,
Hello all,
I want to report a BUG with the Linksys SPA94X so it is general
knowledge and that we can all make noise about it so it will get fixed
sooner..
The handsets do not work with the SIP flag to make them AUTO-ANSWER. (As
documented is should)
Ie, you cannot use them with intercom or
probably it's better to auto-dial the calling phone first, and then let the established channel go out to the recipient!
So when the calling phone answers, the call will go out to the recipient.
Hope this helps...
2006/5/4, Giorgio Incantalupo [EMAIL PROTECTED]:
Hi,I'm using Asterisk 1.2.1 on a
I have thew same problem.
Ui tried with dyn dns in the externip field in sip.conf but I think the
Asterisk does not allow this. Unfortunally I have every day a new ip. Maybe I
can write a script witch takes my actual ip from externat and put it into the
externip field. Maybe this solves the
Hi picciux,
maybe it could work, even if I don't know how to call a phone a channel
and create a bridge between them.
I'd prefer to use Asterisk inner features like the auto-dial out call
moving a .call file to /var/spool/asterisk/outgoing: this works for
analog line but not for ISDN. But only
hi giorgio...
when i said ring the calling phone first I mean using a .call file!
I think now you are doing, in your .call files,something like this:
Channel: Zap/2/3391818250 or mISDN/1/3391818250/s
.
.
.
and the rest to send this channel to the calling phone.
This way, you have to
All sorted now. The features timeout needs to be quite high on mobiles.
After a few tests, it works perfectly.
thanks for your help :-)
Mark
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or
Hi All.
Has there been done any work to support ISAC ?
Thanks,
trond
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
I would like to create an variable length extension that when used with
DISA ends when i dial the pound sign (#) but i cant figure out how to do it
something like
exten = _*21*.# ; but this doesn't work, after i dial # it still waits
a few seconds
I would like to create an variable length extension that when used with
DISA ends when i dial the pound sign (#) but i cant figure out how to do it
something like
exten = _*21*.# ; but this doesn't work, after i dial # it still waits
a few seconds
We have some extensions in our dialplan that start with a star. We can
dial them from Zap phones and SIP phones, but not from phones connected
to a PAP2. After the user presses star follwed by two digits (our
extensions are dialed with star followed by three digits) he hears a
fast-busy that
Hi.
I have a soft phone (X-Lite) which registers with a asterisk server that
can only be accessible once we have some virtual private network
software up and running.
With the above scenario everything works fine.
In the mean time the asterisk server was exposed to the internet, thus
the
Hi List,
Is it possible to store meetme config in a MySQL table?
If so, any pointers would be appreciated.
Thanks
Chris
--
Chris Blunt
Entropy IT Ltd
___
--Bandwidth and Colocation provided by Easynews.com --
In article [EMAIL PROTECTED],
Chris Stenton [EMAIL PROTECTED] wrote:
This is a known problem and it does not matter what zaptel timer you use. A
solution is available in 'svn head' by using
asterisk.conf
internal_timing = yes
OR
Enable internal timing support (-I)
on the command
I think you need to upgrade to the latest Asterisk. Your version is pretty
ancient.
We are using v1.0.8
- Original Message -
From: Michael George [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Thursday, May 04, 2006 2:48 AM
Subject: [Asterisk-Users] meetme conference
I assume you mean this:
http://en.wikipedia.org/wiki/ISAC
but maybe you are referring to one of the controller chips on BRI
adapters?
James
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Trond G. Andersen
Sent: Thursday, 4 May 2006
Title: Messaggio
Under Advanced make sure this is set:
Call join on Xfer (2 calls): to off
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tommaso Calosi
Sent: Thursday, May 04, 2006 4:02
AM
To:
asterisk-users@lists.digium.com
Subject: [Asterisk-Users]
If you have both sides of the call it is possible. It may not be practical
though. If one side was using spandsp then it is both possible and
practical.
Craig
- Original Message -
From: Steve Totaro [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
Hi all,
I have to bought a PCI with 4 PRI but on digium site
I saw that there a re two different kind (3,3V and 5v). Whats the
difference?
Which one I have to buy for do not have any problem
with this motherboard? (Gygabyte GA-8S661FXM-775). I checked on Gigabyte
website but I dont find
cat /proc/interrupts
CPU0 CPU1 CPU2 CPU3
0:422 0 0 46196367IO-APIC-edge timer
8: 0 0 0155IO-APIC-edge rtc
9: 0 0 0 0 IO-APIC-level acpi
14: 414685
5 volt will be for desktop class motherboards and 3.3v for server class.See http://www.digium.com/en/docs/misc/pci_slot.phpRob
On 04/05/06, Giordano Grandis [EMAIL PROTECTED] wrote:
Hi all,
I have to bought a PCI with 4 PRI but on digium site
I saw that there a re two different kind
Hi,what about the web meetme sourceforge project?Has it been approved?benq
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
Hello,
I would like to use an ocx for integrated a softphone in an existant
program developped in Windev (from PC Soft).
I try IaxClientOCx, but nothing happen at initialising. Then, I try some
softphone make with it, it doesn't function either...
Do you know any other OCX for try?
Best
2006/5/3, Asterisk User [EMAIL PROTECTED]:
Does Asterisk support QSIG SIP Tunneling or QSIG SIP Interworking?
Do you mean something like ECMA 336 ?http://www.ecma-international.org/publications/files/ECMA-ST/Ecma-336.pdfRegards
___
--Bandwidth and
2006/5/3, Marco Mouta [EMAIL PROTECTED]:
http://www.voip-info.org/wiki-Asterisk+config+zapata.conf
I've made some tests using this in Portugal and seems to work:---
switchtype=qsig ; you may try this in your
How can you store pauses in speed dials for the GXP-2000? I used
something like 8005551212,,,1,7890 to dial the toll free number, wait
6 seconds (I'm used to the commas being a 2 second delay), pressing
1, waiting 2 more seconds and then entering 7890. However, when I
press the speeddial
2006/5/4, Craig Guy [EMAIL PROTECTED]:
If you have both sides of the call it is possible.It may not be practicalthough.If one side was using spandsp then it is both possible andpractical.CraigCould you elaborate ?
And if a fax is recorded with Asterisk voicemail application (in case an error in
Attribute Values Default Interpretation call.rejectBusyOnDnd 0, 1 1 If set to 1, reject all incoming calls with the reason “busy” if do-not-disturb is enabled. Have not used, but looks like it may ignore the key if this is 0Let us know...On May 4, 2006, at 2:22 AM, [EMAIL PROTECTED] [EMAIL
snip
This is a very KGB / NSA / InterPOL / CIA type
question, but if I have a
recorded file (G.711, no compression) can I feed it
into standard in of
an application and have it recreate the fax that was
send?
/snip
What is the specific reason as to why you want to
record it to a file and
Yes, sorry I was wondering if anyone is working on ISAC voice codec
I have seen a patch
http://lists.digium.com/pipermail/asterisk-commits/2006-February/001461.
html
But not seen anything anywhere else...
trond
I assume you mean this:
http://en.wikipedia.org/wiki/ISAC
but maybe you
QSIG was just the protocol communication between Legaccy PBX and Asterisk.My users connect to Asterisk through SIPOn 5/4/06, Olivier Krief
[EMAIL PROTECTED] wrote:
2006/5/3, Marco Mouta [EMAIL PROTECTED]:
http://www.voip-info.org/wiki-Asterisk+config+zapata.conf
I've made some tests using this
Well, yes and no. I tested that before and it causes a silent ring
instead of a call rejection. I actually want to disable the entire feature. So
the phone always rings unless you're actually on the phone.
Thanks for the reply though!
Regards,Jan
Från: [EMAIL PROTECTED]
[mailto:[EMAIL
In the PAP2's setup there are all of these Vertical Service Activation
Codes that start with star and Outbound Call Codec Selection Codes,
also the setup menu is accessed by pressing star four times, could they
be intefering with dialing numbers that start with a star? And is there
any way to
I have a client that 'NEEDS' (his words not mine) to make sure that all
faxes, emails, calls, and mail are archived. Phone and email are simple,
Mail depends upon the integrity of the mail room, Faxes however can be
sent from anyone. They would like this as they recently had an issue
with a fax
Hi all,
I am trying to detect DTMF keys from a mobile when asterisk make an
outgoing call to the mobile.
The DTMF detection on incoming calls (also FROM mobiles) is working very
well.
The only problem is if asterisk called the phone... Nothing is detected.
I am using a digium te205p with
From my recent problem on this sort of thing, I'd suggest you set the
timeout to around 1500ms in the feature.conf file. This is of course if your
using the DTMF digit's to activate any of the features.
also make the devices both sides of the call are using the same DTMF mode.
Mark
Hi all,
Marc Scheuffler wrote:
Hi all,
I am trying to detect DTMF keys from a mobile when asterisk make an
outgoing call to the mobile.
The DTMF detection on incoming calls (also FROM mobiles) is working very
well.
The only problem is if asterisk called the phone... Nothing is detected.
I am using a
Yes I can hear the DTMF keys. I ve tried 2 different phones and 3 different
mobile network providers. Nothing.
I played with the rx/txgain values from hearing nothing to too loud...
I have no more ideas.
Marc
-Ursprüngliche Nachricht-
Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
Before I go nuts trying to figure this out, is anyone using DISA in this manner?
exten = s,1,DISA(X|context|callerid)
Everything works except the caller ID part. What I had wanted to do is to setup up a file of authorization codes where each code was associated with a context and caller id.
Yapp, timeout is set to 1500ms.
What kind of dtmf mode? As far as i know there are just 2.
Relaxdtmf yes or no
Or am I wrong?
-Ursprüngliche Nachricht-
Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Im Auftrag von Mark Ackroyd
Gesendet: Donnerstag, 4. Mai 2006 16:52
An: 'Asterisk
In the PAP2's setup there are all of these "Vertical Service Activation Codes" that start with star and "Outbound Call Codec Selection Codes", also the setup menu is accessed by pressing star four times, could they be intefering with dialing numbers that start with a star? And is there any
Just wanted to add my 2 cents. We were with voipjet, and do still use
them for occassional backup.However, their lack of personal
service and inability to get ahold of someone drove us away.After
several total blackouts (like what happened yesterday), and no
responce we finally put out
I've got this low-ping 100%-up dsl connection between two asterisk
1.2.7.1 servers. And oftentimes one of them would declare its opposite
UNREACHABLE.
Why can this happen? The host stanzas in iax.conf have raw IP's, so no
DNS monkey business here.. An inquiring mind wants to know.
I have tried using the autologoff in the agents.conf and it sort of works. I
set it to 5 seconds to test it and it has taken anywhere from 35 to 60
seconds to actually do something at which point it does indeed log out the
agent.
I don't want to be pestering agents with test calls to see if they
Why not capture the faxes (in or out) in tiff format, instead of audio
format? Setup your asterisk box to relay faxes!
MD
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alexander
Lopez
Sent: Thursday, May 04, 2006 10:07 AM
To: Asterisk Users Mailing
On Thursday 04 May 2006 11:31, Louis-David Mitterrand wrote:
I've got this low-ping 100%-up dsl connection between two asterisk
1.2.7.1 servers. And oftentimes one of them would declare its opposite
UNREACHABLE.
I see this happen on occasion as well -- same type of setup here, static IPs,
no
Hi,
Did anybody succeed remapping soft-keys on polycom 301 ?
I am having some problems with it.
I was trying to remap Transfer button as the first option while being in a
call.
It works but The name of the soft key is still HOLD and while I am not
in a call I see button NewCall that
It has been approved. We started out trying to use
CVS on SourceForge, but
it appears that there have been major issues with CVS, so
we just switched to
SVN.
We need to
checkin a baseline, and start integrating patches.
Dan
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
No. iSAC is a codec from GIPS. Likely the coded used by Skype.
Michael
On Thu, 4 May 2006 21:35:07 +1000, James Harper wrote:
I assume you mean this:
http://en.wikipedia.org/wiki/ISAC
but maybe you are referring to one of the controller chips on BRI
adapters?
James
-Original
Why not capture the faxes (in or out) in tiff format, instead of audio
format? Setup your asterisk box to relay faxes!
I think in this case the impact on the client would be much greater if you
can show them a recreation of the image from the raw data; you could always
claim that a TIFF file was
Andrew Kohlsmith wrote:
On Thursday 04 May 2006 11:31, Louis-David Mitterrand wrote:
I've got this low-ping 100%-up dsl connection between two asterisk
1.2.7.1 servers. And oftentimes one of them would declare its opposite
UNREACHABLE.
Same, here, two asterisk 1.2.7.1 boxes connected to the
Colin Anderson wrote:
Why not capture the faxes (in or out) in tiff format, instead of audio
format? Setup your asterisk box to relay faxes!
I think in this case the impact on the client would be much greater if you
can show them a recreation of the image from the raw data; you could
http://www.soonr.com/web/front/features.jsp
Just saw this on the Always On Top 100 webcast (if you arent
familiar click the url below)
http://deancollinsblog.blogspot.com/2006/05/always-on-awards-top-100-of-2006.html
Soonr looks like it rocks, havent tried it yet.
Cheers,
Anyone use this thing?
http://www.dlink.com/products/?pid=426
The fab sheet is totally useless for tech info. How does it work? By ToS?
Port number? Is it programmable? Can I prioritize an arbitrary port or ToS
bit?
tia
___
--Bandwidth and
I am trying to use QSIG to interoperate with legacy PBXs.
I am looking to see whether any one knows whether Call Hold, Call Transfer, MWI
works with QSIG support in Asterisk.
Thanks in advance.
--Pillai
On 5/4/06, Olivier Krief [EMAIL PROTECTED] wrote:
2006/5/3, Marco Mouta [EMAIL
'recognize'? The phone cannot know that the external IP has
been changed, unless it is using a STUN server and
periodically re-doing the STUN queries (which I doubt any phones do).
Thanks for clearing up my misunderstanding as to the point of STUN. :-) I
thought the phone would query the
I would have no problem decoding a FAX, doctoring the images, then
creating modified audio from them. During decoding, the FAX modems
produce a channel estimate, so reproducing the characteristics of the
original audio path wouldn't be hard. I think it would be pretty easy to
create fresh
Can I mix these in a single system... having problems getting
the tor2 driver or the wct4xxp drivers to load, although they
seem fine if alone in the system.
span=1,0,0,esf,b8zsbchan=1-23dchan=24
span=2,0,0,esf,b8zsbchan=25-47dchan=48
span=3,0,0,esf,b8zsbchan=49-71dchan=72
On Thursday 04 May 2006 14:08, Steve Underwood wrote:
I would have no problem decoding a FAX, doctoring the images, then
creating modified audio from them. During decoding, the FAX modems
produce a channel estimate, so reproducing the characteristics of the
original audio path wouldn't be
I am getting read to roll out close to 100 polycom phones and wondered if any one knows of a program to take a list of MAC addresses, extensions, and names and generate the configuration files?-- Bruce
Nortex Networks
___
--Bandwidth and Colocation
Colin Anderson [EMAIL PROTECTED] writes:
Why not capture the faxes (in or out) in tiff format, instead of audio
format? Setup your asterisk box to relay faxes!
I think in this case the impact on the client would be much greater if you
can show them a recreation of the image from the raw data;
All, this doesn't appear normal to me, it appears as if ast is ignoring
the itignoreexpire variable.
sip.conf snippet:
rtignoreexpire=yes
asterisk -r
CLIsip show settings
--snip--
Ignore Reg. Expire: No
--snip--
Does this look like a problem? :-)
Thanks, Matt
Hi there,
I'm looking into developing an in-house switchboard application. Does
anyone here know of a way to control a hard-phone from such an
application.
For example, the attendant forwards a call with another one in queue.
Once the first call has been forwarded (by keyboard shortcuts or
On Thursday 04 May 2006 14:45, Bruce Reeves wrote:
I am getting read to roll out close to 100 polycom phones and wondered if
any one knows of a program to take a list of MAC addresses, extensions, and
names and generate the configuration files?
You can do this relatively easily with Perl.
Why is this hard to fake at all? You send a different fax to your
system, and replace the Asterisk audio file with the one from the
altered fax. Additionally, the client has no realistic way of
verifying the correctness of your audio-to-fax translation tool; it
could just as easily output a TIFF
Try this one:
http://www.freedomphones.net/polycom/files/polycom.phone1cfg.pl-script
Sean
Andrew Kohlsmith wrote:
On Thursday 04 May 2006 14:45, Bruce Reeves wrote:
I am getting read to roll out close to 100 polycom phones and wondered if
any one knows of a program to take a list of MAC
Something I made might help.
http://www.horanappraisals.com/asterisk/polycom_addphone/ -- there is a
script, addphone, and a folder called defaults that contains the
templates.
To use, I put the defaults folder and its contents and the addphone
script in my ftp or tftp root. I would make
Hi Bruce,
We've written software to do this as a service for our
customers. I can't give you the program, but we'd be willing to program
your phones for you. Contact me off list.
Michael Crown Managing Partner
www.thevoipconnection.com 321.989.6728 ext. 611 sip:[EMAIL PROTECTED]
Hard to believe you arent associated with calleveryone.com as I find it hard
to believe that you would be extolling the virtues on one of, if not the
most expensive companies around. $7 a month plus 3.9 cents a minute
domestic, that's pretty much double the cost of anyone else. Customer
service
Sounds like a potential business opportunity. Someone could setup a fax
proxy service that provides this sort of digital signing / archiving.
The originator could simply dial a toll-free access number, receive a
2nd dialtone and then dial the destination. Meanwhile the proxy is
recording the call,
On Thursday 04 May 2006 15:18, Sean Cook wrote:
http://www.freedomphones.net/polycom/files/polycom.phone1cfg.pl-script
Yep that's the one that reads sip.conf and spits out phone[exten].cfg files.
It does not tie in mac addresses nor generate [macaddress].cfg files, though.
-A.
it is two scripts an empty_queue.sh and a fill_queue.sh and a members script If you need intructions please tell me1047 $ cat empty_queue.sh#!/bin/bash# a script to remove everyone in the members script located in the same directory as this file
# to the Q 3901# can be called from a
Colin Anderson [EMAIL PROTECTED] writes:
Why is this hard to fake at all? You send a different fax to your
system, and replace the Asterisk audio file with the one from the
altered fax. Additionally, the client has no realistic way of
verifying the correctness of your audio-to-fax translation
Kerry,
You didn't read my entire e-mail. How do I know that? Because if
you re-read it you'll see that I state:
If you are a wholesole buyer of minutes, talk to them, don't just
take their prices on the main page... those are for residential and
regular customers. Their prices are very
We are using the polycom 501 phones, and are having some challengeswith the volume setting. When a phone call comes in, the user ups thevolume for the handset, but they have to repeat that for every call.Currently, the volume level seems to reset itself at about 60%.
Is there a way for the user to
--- Vahan Yerkanian [EMAIL PROTECTED] wrote:
Andrew Kohlsmith wrote:
On Thursday 04 May 2006 11:31, Louis-David
Mitterrand wrote:
I've got this low-ping 100%-up dsl connection
between two asterisk
1.2.7.1 servers. And oftentimes one of them would
declare its opposite
UNREACHABLE.
Hrmmm.
I thought there was already an option in the queue.conf or agents.conf
file (Though can't remember off hand what) that would set an agent
logged out or on 'pause' if they did not answer a call. No?
On 5/4/06, Christopher Mayfield [EMAIL PROTECTED] wrote:
it is two scripts an
Hi Bruce,
We create a CSV file of our phone setup and then use shell scripts to
parse them and generate mac-address.cfg, phone.cfg, sip.conf,
voicemail.conf and entensions.conf entries.
Contact me off list if you would like a copy now (they're not quite
ready for prime-time yet) - the
That is what I thought too, but what about this:
http://lists.digium.com/pipermail/asterisk-commits/2006-February/001461.
html
???
No. iSAC is a codec from GIPS. Likely the coded used by Skype.
Michael
On Thu, 4 May 2006 21:35:07 +1000, James
sip.cfg
volume voice.volume.persist.handset=1
voice.volume.persist.headset=1 voice.volume.persist.handsfree=1/
Jim Freeze wrote:
We are using the polycom 501 phones, and are having some challenges
with the volume setting. When a phone call comes in, the user ups the
volume for the handset,
This has got to be a stupid error I'm making...
I have been experimenting with different hardware and software
configurations before I decide what to use as a production platform. Up
until just recently things were going well.
But now it appears I'm unable to get access to my TDM400p from
Title: Message
I have a conflict problem with the eth0
card and wct2xxp digium board. The PRI can
receive calls but my network connection is gone.
When I "cat /proc/interrupts" I get the
following:
1
..
1 ..
..
..
..
169 0 IO-APIC-level wct2xxp,
eth0
..
etc.
even before I "modprobe
Edit your config files to enable persistance
Will remain across multiple calls, but not reboots
On May 4, 2006, at 2:51 PM, Jim Freeze wrote:
We are using the polycom 501 phones, and are having some challenges
with the volume setting. When a phone call comes in, the user ups the
volume for
You can use my script, based on Chris Mason's script, to do most of what
you want, you can feed it your MAC's and Extensions and it will create the
phones.
Be warned, it's not pretty, my perl book was in storage so I did a lot of
kludging. Feel fee to update.
On Thursday 04 May 2006 15:51, Tom Engleward wrote:
Am I supposed to make a cron job to automatically tell
asterisk to reload every so often, since iax2 likes to
periodically die? Or maybe am I supposed to make a
cron job to place a phone call every so often from an
external phone into my
couple of things... was asterisk compiled after zaptel? from the cli
try load chan_zap.so and see what you get
Ben Gore wrote:
This has got to be a stupid error I'm making...
I have been experimenting with different hardware and software
configurations before I decide what to use as a
I have asterisk
1.2.7.1 running on Fedora core 5. Everything looked like it compiled
OK.
When a call is
bumped to voicemail, the message prompts sound fine to the user. However,
when thevoice message is retrieved, it sounds "compressed" or speeded
up.
I have checked this
against the
try http://sourceforge.net/projects/web-meetmeChris Blunt [EMAIL PROTECTED] wrote:Hi List, Is it possible to store meetme config in a MySQL table?If so, any pointers would be appreciated.ThanksChris --Chris Blunt Entropy IT Ltd
Are you specifying the remote Asterisk box by IP or by hostname. If by
hostname, then specify it by IP. Asterisk's DNS lookup support has issues.
2) What is your qualify= set to. Set it to yes (2000), or don't set
it at all. Also look at the qualify smoothing options in iax.conf.sample.
1 - 100 of 132 matches
Mail list logo
