Perhaps it's a codec issue.
Which codec are you using ulaw, alaw, gsm ...?
Try to use the same codec on every phone and allow that codec explicit in
sip.conf .
Guido Hecken
-Ursprüngliche Nachricht-
Von: Voip Business [mailto:[EMAIL PROTECTED]
Gesendet: Donnerstag, 24. Februar 2005
Paul P. Pongco wrote:
Hello List,
Can you please point me to the right resources on making multiple sip
phones behind a firewall w/ private address work with asterisk w/c is on
a public network.
I have seen STUN on the grandstream and Xtunnels on X-lite. What is most
deployed by members here with
Hi
I was wondering if there are any special settings that
I need to be able to transfer calls.
Whenever I press the 'recall' button, I just here a click,
and no ring-tone to transfer.
in my debug log I get this :
--
Feb 24 09:09:27 DEBUG[19216]: Exception on 10,
On Thu, 24 Feb 2005, Kuniyoshi Murata wrote:
I want to have a single meetme conference room that interconnects H.323
video phone clients and sip/iax audio phone clients.
I have already set up for meetme to be shared by sip/iax audio phones and I
have just now installed open h323 stuff.
That was my impression as well, I tried adding switchtype=qsig in zapata.conf,
but all I see is the capi information... No connection...
Janne
-Ursprungligt meddelande-
Från: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] För Peter Svensson
Skickat: den 23 februari 2005 21:34
Till:
Fair point. The FAX apps used to display a note about this when you
checked their parameters. That seems to have disappeared from the source
code of both rxfax and txfax. I just put it back in.
Regards,
Steve
Rod Bacon wrote:
From my personal experience, the 'weird ideas' come from a lack of
I tried to force the modules (man insmod = modprobe -f in Mandrake
land?) and zaptel looked good, unfortunately wcfxo gave the same errors
as ever.
I have modified the /usr/src/linux/makefile by removing the word custom
from the end of the EXTRAVERION statement, so it now looks like -
VERSION
On Thu, 24 Feb 2005, Jan Berggren wrote:
Is the zapata.conf file used at all for CAPI? I though all
the physical to connection layer stuff was handled by capi
and not asterisk? Wouldn't the signalling be configured in
the capi configuration files?
That was my impression as well, I
Hi all,
I have call recording enabled via the Monitor command and it seems,
the call stops being recorded after the call is transferred. Is this
normal behavior? If so how can I continue recording of calls after
they have been transferred
___
I have read most of Eicons information on Q.SIG, and I am able to load the
Q.SIG protocol (instead of ETSI for example). No strange logging in divacrtl
mlog.
But how do I tell Asterisk to understand Q.SIG?
My PBX is configured for QSIG, but I cannot see anything on my trace when
trying to
All,
I have downloaded and installed openh323 as per the documentation.
When the machine now reboots safe_asterisk just keeps restarting.
If I start another session and just load asterisk -vvvgc asterisk loads.
If I enter noload chan_h323.so in the modules.conf then safe_asterisk will
kick
Hi,
I have just recently started using an analogue telephone extension and
was wondering what the hold key sequence is. I can use # to transfer a
call (sometimes) but what do I do if I just want to place a caller on
hold and then retrieve them (for example I want to have a private
conversation
Hello,
I have installed asterisk but it is impossible to get compiled the
chan_capi 0.3.5.
I am using slackware 10.1 (kernel 2.4.29) and I am getting the following
when I issue gcc -v
Reading specs from /usr/lib/gcc-lib/i486-slackware-linux/3.3.4/specs
Configured with: ../gcc-3.3.4/configure
On Thu, 24 Feb 2005, Jan Berggren wrote:
I have read most of Eicons information on Q.SIG, and I am able to load
the Q.SIG protocol (instead of ETSI for example). No strange logging in
divacrtl mlog.
But how do I tell Asterisk to understand Q.SIG?
Is Asterisk involved on a low enough level
Hi to all
my local reseller gave me this price for the Eicon DIVA server boards...
Diva Server BRI-2M 749 Euros
Diva Server 4BRI-8M ..1927 Euros
Diva Server PRI E1/T1 3796 Euros
I think that they are expensive. Is this the normal price?
I just hope that Asterisk and my
On Wed, Feb 23, 2005, Scott Stingel wrote:
Note that a CAT5 crossover cable will not work. Once you've done this,
set up two spans on your TE410P as you've done it, except that one of
them should be CPE and the other one NET in zapata.conf. Now calls
originated on any of the 30 channels
Hello,
I had a major headache trying to send faxes for 2 weeks now.
Two things helped me:
1. Correct TIF files (received file not always worked for me)
2. Time synchronization e.g. 'ntpdate time.windows.com'.
Now %90 of faxes come nice and clean. (10% - incomplete).
If you want test TIF file
Hi @all,
I have a question concerning bridging calls in asterisk.
Person A recieves a call fom Person X. Person X wants to talk to Person
B. Person A starts contacting person B and introduces person X.
Person B now transfers the call and person X is directly connected to
person B.
Is it
Do I have to cd into my asterisk source directory (that is,
/usr/local/etc/asterisk) or otherwise?
Yes, cd in your asterisk source dir, from where you installed it.
Secondly, is the statement no.2 a line a need to change in a given file?
You have to change/verify some settings in
I get the impression that the transfer/flash/recall etc etc buttons
don't always work - it seems to depend on what phone/firmware you are
using. And possibly the version of asterisk.
I am using BT102s and some generic voip phone. On the BT102 the transfer
button will put the call on hold and
Do you mind sharing where in the configs you have changed this ?
-Herman
On Thu, 2005-02-24 at 12:40, Mark Benson wrote:
I get the impression that the transfer/flash/recall etc etc buttons
don't always work - it seems to depend on what phone/firmware you are
using. And possibly the
I do follow all the instructions except for one; I don't seem to have the
phpconfig_init.php file in my asterisk source directory. Maybe I have
overlooked something or forgotten to install a certain package.
Do I need to install some package that has the phpconfig_inti.php file?
How do I go about
Are you using a Diva SERVER board or just a Diva PCI?
I also whant to connect Asterisk with a Siemens HH3000, but I whant to know
if it can be done with an Eicon PCI or with a Digium board, because the
Eicon DIVA Server 4BRI is very expensive.
Joao Pereira
- Original Message -
From:
Herman
Do you mind sharing where in the configs you have changed this ?
Have a look at the options that are controlled by settings in
features.conf. If these are what you want to control, I think you will
find the options obvious.
Bill Seddon
Lyquidity Solutions Limited
-Original
On Wed, 2005-02-23 at 20:32, Nabeel Jafferali wrote:
Does AreskiCC allow the card number to be passed from the dialplan, like
ASTCC does?
Yes!
You can use SetAccount(cardnumber) before the DeadAGI or you
can also define the accountcode = cardnumber into the sip.conf or
iax.conf.
Regards,
On Thu, 2005-02-24 at 19:57 +1100, Eric Bishop wrote:
Hi all,
I have call recording enabled via the Monitor command and it seems,
the call stops being recorded after the call is transferred. Is this
normal behavior? If so how can I continue recording of calls after
they have been
We've been using SIP with Asterisk for a couple of years now, and it's
generally worked fine. However we're now trying to use a more
complicated codec setup, and I've hit a problem with how codecs are
selected that I can't get around.
For a simple configuration:
XLite GSM Asterisk
where GSM
On Thu, 2005-02-24 at 10:20 +0200, Herman Cremer wrote:
Hi
I was wondering if there are any special settings that
I need to be able to transfer calls.
Whenever I press the 'recall' button, I just here a click,
and no ring-tone to transfer.
in my debug log I get this :
Probably your
On Thu, 24 Feb 2005, Joao Pereira wrote:
my local reseller gave me this price for the Eicon DIVA server boards...
Diva Server BRI-2M 749 Euros
Diva Server 4BRI-8M ..1927 Euros
Diva Server PRI E1/T1 3796 Euros
I think that they are expensive. Is this the normal
Hi Kristian,
Anywhere I can read about this Soekris/AstLinux project? ...
Regards,
Hans
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Kristian
Kielhofner
Sent: Thursday, February 24, 2005 6:02 AM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List -
i want to use the call monitoring for my litttle
exchange. As my requirements are to monitor each and every call. Is there any
way to do this wil a single command or i shall have to write it for each and
every extension and the incomings and the outgoings.
How about getting rid of the drives (hard, floppy, cd, dvd) and using
RamDisk technology instead. You can boot from flash memory. This will
reduce heat and increase reliability.
Jonathan
- Original Message -
From: Vledder, Hans [EMAIL PROTECTED]
To: Asterisk Users Mailing List -
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Mark Benson
Sent: Thursday, 24 February 2005 8:41 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] CallTransfer
I get the impression that the
Nathan C. Smith wrote:
I have InAccess Networks' oh323 installed and partially working. I can call
the h.323 phone from asterisk using Dial(oh323/${IP_ADDRESS}). How do I
dial from the phone to an asterisk extension? It does not appear to me that
the phone actually registers (or attempts to
-Original Message-
From: Nic le Roux [mailto:[EMAIL PROTECTED]
Sent: 24 February 2005 12:39 PM
To: 'Julian J. M.'; 'Asterisk Users Mailing List - Non-Commercial
Discussion'
Subject: RE: [Asterisk-Users] SIP echo on LAN
Do you mean that I need to check the sound card settings on the
On Thu, Feb 24, 2005, Johan Bilien wrote:
Note that a CAT5 crossover cable will not work. Once you've done this,
set up two spans on your TE410P as you've done it, except that one of
them should be CPE and the other one NET in zapata.conf. Now calls
originated on any of the 30
I am working on call monitoring.
The filename convention that i am using
is:
${EXTEN}--- timestamp.wav
i want to add the callers extension to it as well.
Is there any variable alreadu defined for that?
Kindest
Muhamamd Muzzamil Luqman
___
Hello:
I have an asterisk deployment with 15 MGCP extensions and 30 incoming
E1 R2 channels. Calls are received by a receptionist queue (which only
member is the receptionist phone). The receptionist then transfers
(using hook flash) the call to one of the extensions. I want to be
able to have
This is also installed automatically using the [EMAIL PROTECTED] solution.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Hecken, Guido
Sent: Thursday, February 24, 2005 5:42 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE:
hah !
worked like a bomb ..
Thanks a million.
-Herman
On Thu, 2005-02-24 at 13:49, Adam Goryachev wrote:
On Thu, 2005-02-24 at 10:20 +0200, Herman Cremer wrote:
Hi
I was wondering if there are any special settings that
I need to be able to transfer calls.
Whenever I press the
Hello All
I am wondering is someone knows how to configure the * to work with an
Ericsson MD-110 with SL60 signaling?? through a TLU76 card. What is the
right configuration in the zaptel.conf ? I currently have it configured
as span=3.0.0,ccs,hdb3,crc4 but it doesn't detect anything when I
[EMAIL PROTECTED] is believed to have said:
I am using BT102s and some generic voip phone. On the BT102 the transfer
button will put the call on hold and give you a new line to call an
extention with, however nothing happens when I call an extention. On the
generic voip phone the transfer
[EMAIL PROTECTED] is believed to have said:
I am using slackware 10.1 (kernel 2.4.29) and I am getting the following
when I issue gcc -v
Dimitris,
while I never compiled chan_capi I thought you would need a 2.6 kernel to
use it.
HTH
Aldo
___
On Thu, 24 Feb 2005, Johan Bilien wrote:
On the latter, I can choose the linetype to be one of the following:
E1Unframed - 2 Mbit/sec unframed signal
E1 - 2 Mbit/sec framed PCM31 signal
E1Crc- 2 Mbit/sec framed PCM31 signal with CRC
E1Mf
Yep, works for me to. I'm using revision 1.0.5.22 software and the
[EMAIL PROTECTED] solution.
I think it has worked since revision .16
Cheers,
Dean
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Aldo
Bergamini
Sent: Thursday, February 24, 2005 7:55
I agree with peter stick to digium cards at least you'll have native
support for asterisk.
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
On Thu, 24 Feb 2005 19:13:27 +1100, David Uzzell wrote:
Paul P. Pongco wrote:
Hello List,
Can you please point me to the right resources on making multiple sip
phones behind a firewall w/ private address work with asterisk w/c is on
a public network.
I have seen STUN on the grandstream
Hi,
Does anybody has experience with high capacity PSTN voicemail and
asterisk, running more then 5k mailboxes for PSTN users ?
How many mailboxes can I serve with 4xE1 card if we assume that we
have enough harddrive
capacity. What would be server requirements. Would the CPU load be the
same when
Aldo Bergamini wrote:
I am using slackware 10.1 (kernel 2.4.29) and I am getting the following
when I issue gcc -v
while I never compiled chan_capi I thought you would need a 2.6 kernel to
use it.
You don't need a 2.6 kernel for chan_capi.
--
Best regards
Peer Oliver Schmidt
PGP Key ID:
Hi all
I'm fairly new to Asterisk, so be nice :-)
I was wondering if anyone has been able to get the Welltech K1000A USB
phone working on Linux. I see audio and HID drivers loaded when it is
plugged in to my Fedora Core 1 laptop, but that's about all that
happens. I've searched all the usual
I noticed that the Aastra 480i has a telnet interface, but the login is
different than the web. Anyone know how to log into the telnet
interface? Is the interface documented anywhere?
Thanks,
Philip
___
Asterisk-Users mailing list
Philip Trauring wrote:
I noticed that the Aastra 480i has a telnet interface, but the login is
different than the web. Anyone know how to log into the telnet
interface? Is the interface documented anywhere?
You need to use a telnet client program to communicate with the device.
Under Windows
-Ursprungligt meddelande-
Från: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] För Peter Svensson
Skickat: den 24 februari 2005 10:43
Till: Asterisk Users Mailing List - Non-Commercial Discussion
Ämne: Re: SV: SV: [Asterisk-Users] QSIG, Asterisk and Eicon DIVA
On Thu, 24 Feb
I do follow all the instructions except for one; I don't seem to have the
phpconfig_init.php file in my asterisk source directory. Maybe I have
overlooked something or forgotten to install a certain package.
If you get the sources with cvs checkout phpconfig, you should have this
file.
I
Your span definition should be fine (except there should be commas
instead of dots, but that is probably just a typo). You need to play
with various parameters on the MD-110 side, those in RODAI command, as
well as SIG parameter in ROCAI command. I don't know how well is QSIG
implemented in
does MD 110 support SIP?
On Thu, 24 Feb 2005 15:08:49 +, Niksa Baldun [EMAIL PROTECTED] wrote:
Your span definition should be fine (except there should be commas
instead of dots, but that is probably just a typo). You need to play
with various parameters on the MD-110 side, those in RODAI
Hi All,
I have no experience with VOIP or SIP, but I am a *nix networking admin for
some years now and I would like to get into working with SIP etc...
I have installed asterisk and configured it to run with 2 software phones.
This works fine, and I can chat between 2 client machines in our
Do you think create a VideoMail is doable with Asterisk?
Is it possible to create a kind of VideoMail application, for example by
using AGI (or Asterisk Application) to play video files instead of voice
files ?
Any ideas how we can go further with this...
Areski
I'm only just getting my head round asterisk - so the phones themselves
have taken a back seat - I have only recently upgraded the phones to r
.16 - so maybe they do work now. I'll test as soon as all my users have
moved their phones over to the asterisk server.
I only found out about the r
On Thu, Feb 24, 2005, Peter Svensson wrote:
On Thu, 24 Feb 2005, Johan Bilien wrote:
On the latter, I can choose the linetype to be one of the following:
E1Unframed - 2 Mbit/sec unframed signal
E1 - 2 Mbit/sec framed PCM31 signal
E1Crc- 2
Hi,
I got this device and it looks like I can get this to work as a router
connecting my local net to my DSL provider with QoS in place of my old
router.
I am thinking of using it as follows:
Dsl - 2100 - switch with * box and voip phones attached - old router -
other pcs
Does it sound like
Sounds like a great idea and has a heap of commercial applications.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Areski
Sent: Thursday, February 24, 2005 9:19 AM
To: Asterisk-Users Mailing-list
Subject: [Asterisk-Users] VideoMail Asterisk
Do you
Grandstream write the software but for reason anything later than the
1.0.5.16 has only been available fro the betatest section.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mark
Benson
Sent: Thursday, February 24, 2005 9:29 AM
To: Asterisk Users
While trying to
deploy a bunch of Polycom IP 500 phones, I ran in to the following. I
limited the RTP ports from 8000-8050 to limit holes in firewall. Pretty
soon Asterisk ran out of RTP ports. Traced the problem back to how * is
handling SUBSCRIBE. A sip structure is allocated as soon as
Good day,
I am currently taking a look at queues. What I am trying to
achieve is that, beside the MoH, when the caller gets put
into the queue, she should hear an announcement like welcome
to snakeoil - please wait or leave a message by pressing #
then the MoH. The announcement should be repeatet
Thanks for the reply.
did you try the auto-config? help-aah will tell you
how to do this.
Yes I did. It put the span definitions etc. in zaptel.conf *as comments*
(BTW, is this to be expected - that the defenitions are commented?), but if
I uncomment those lines then the it starts giving me
I'm trying to set a channel variable and make it available to another
channel:
I thought that if I SetVar(_SomeVariable=SomeValue) or
SetVar(__SomeVariable=SomeValue) then SomeVariable would be available in
the destination channel.
However __SomeVariable, _SomeVariable and SomeVariable are
What you want is SetGlobalVar, which sets a variable that is available to
any channel.
hth
Jonathan
- Original Message -
From: Asterisk [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: February 24, 2005 10:31 AM
I am having a problem with the Call Queue Feature.
If I let a user into the Queue prior to an agent being available for
them to take the call, they experience the following:
1) they hear that they are the first in line
2) when the agent finally logs in (the caller on hold in the
Jonathan Hobbs wrote:
What you want is SetGlobalVar, which sets a variable that is available to
any channel.
No, I only want the variable available to the original channel and all
connected channels. I don't want it available to any channel.
For example, I want to store the zap channel number
TC wrote:
Any one know of software that allows 2-way radios as VoIP(SIP) clients,
besides dingotel's usb mic cable trick ?
http://www.dingotel.com/2way/requirements2way.asp
They might be ok if the SIP client was not hardcode to their own SIP proxy
Has anyone tried any hacks to get the 2-way radio
Has anyone gotten the call parking soft button on the polycom
phones(specifically the IP 600) to work with call parking that asterisk
provides? If so what configuration changes were needed?
--johann
___
Asterisk-Users mailing list
On Thu, 24 Feb 2005, Asterisk wrote:
I'm trying to set a channel variable and make it available to another
channel:
I thought that if I SetVar(_SomeVariable=SomeValue) or
SetVar(__SomeVariable=SomeValue) then SomeVariable would be available in
the destination channel.
However
On Thu, 24 Feb 2005, Sascha E. Pollok wrote:
I am currently taking a look at queues. What I am trying to
achieve is that, beside the MoH, when the caller gets put
into the queue, she should hear an announcement like welcome
to snakeoil - please wait or leave a message by pressing #
then the
Nop
does MD 110 support SIP?
On Thu, 24 Feb 2005 15:08:49 +, Niksa Baldun niksa.baldun at
lumiss.hr wrote:
Your span definition should be fine (except there should be commas
instead of dots, but that is probably just a typo). You need to play
with various parameters on the MD-110 side,
Julian
For example, I want to store the zap channel number into
SourceChannel
on the incoming call, and make it available to the called agent
channel.
If global variables are not suitable maybe the registry will work. You
may have looked at DBGet and DBPut to retrieve and store arbitrary
Hmm. I'm running
Asterisk CVS-HEAD-02/16/05-07:03:04
I'll look into the bug list.
Julian.
Peter Svensson wrote:
On Thu, 24 Feb 2005, Asterisk wrote:
I'm trying to set a channel variable and make it available to another
channel:
I thought that if I SetVar(_SomeVariable=SomeValue) or
Just to clarify, I know how to use telnet - what I'm looking for is the
username/password pair I need to access the phone via telnet, and
hopefully some documentation on what I can access via telnet. I am
particularly interested in using the interface to reset the phone
remotely.
Philip
On
Bill Seddon wrote:
Julian
For example, I want to store the zap channel number into
SourceChannel
on the incoming call, and make it available to the called agent
channel.
If global variables are not suitable maybe the registry will work. You
may have looked at DBGet and DBPut to retrieve and
I tried getting this to work and gave up when the polycom admin guide said the park button didn't work in SIP or MGCP applications. Don't know why it's there.
On Thu, 2005-02-24 at 09:54, Johann wrote:
Has anyone gotten the call parking soft button on the polycom
phones(specifically the IP
Jonathan Hobbs wrote:
How about getting rid of the drives (hard, floppy, cd, dvd) and using
RamDisk technology instead. You can boot from flash memory. This will
reduce heat and increase reliability.
Jonathan
That's exactly what I'm doing. I boot a SBC from flash that doesn't
have fans or even
Vledder, Hans wrote:
Hi Kristian,
Anywhere I can read about this Soekris/AstLinux project? ...
Regards,
Hans
http://www.soekris.com
http://www.kriscompanies.com/modules.php?name=Contentpa=showpagepid=1
--
Kristian Kielhofner
___
Asterisk-Users mailing
Sarat Vemuri wrote:
While trying to deploy a bunch of Polycom IP 500 phones, I ran in to the
following. I limited the RTP ports from 8000-8050 to limit holes in
firewall. Pretty soon Asterisk ran out of RTP ports. Traced the
problem back to how * is handling SUBSCRIBE. A sip structure is
Pessoal estou querendo montar um servidor
SIPpara fazer testes, alguem me recomenda um stavel e com interface
web para que se vcs quizerem tambem o usem para testes?
Max Rivera
Megaweb IDC
___
Asterisk-Users mailing list
On Thursday, 24 February, 2005 12:54 : Peter Svensson [EMAIL PROTECTED]
wrote:
The Diva Server cards are expensive because they have on board dsp
chips. Have you considered the cheaper alternative of using Junghanns
quad/octobri cards for bri and Digium TE4xxP cards for PRI? Both of
these use
I have been modifying settings in /usr/src/linux/makefile and if I
modify this to -
VERSION = 2
PATCHLEVEL = 6
SUBLEVEL = 8
EXTRAVERSION = .1-12mdk-i586-up-1GB
I get the following in my /var/log/messages file -
Feb 24 16:08:19 asterisk kernel: zaptel: version magic
'2.6.8.1-12mdk-i586-up-1GB
*The Asterisk http://www.asteriskpbx.org app_rpt project
The integration of 2-way radio systems and reasonable telephony
*http://www.zapatatelephony.org/app_rpt.html
do you know if its possible to interface the analog telephony card
using the 9 pin connector to the mic-in jack of a motorola
do i have to reload asterisk every thing i add a new extension
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
I hae tried searching the web for the answer, but, man is there a lot of
pages ... :(
in the language I develop in, if I have a structure I can dynamically
refer to the contents of a field of the structure like so:
MESSAGE SomeStructure:Field(SomeFieldName):Value
where SomeFieldName is
Kanishka Somaratne wrote:
do i have to reload asterisk every thing i add a new extension
try extensions reload from the CLI
Julian
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To
Problem: Zap Channels Disappear @ random intervals. (Channels
have disappeared on both gateways twice this week).
My Setup:
I have 2 Dell 1850 Power Edge Servers Configured as.
P4 2.8Ghz
512 ECC Memory
SCSI Array (2 Drives Mirror)
Configuration is really simple the boxes are
Hi there,
Do anyone have any experience with SIP phone that support
a headset ? We have Budgetone phones but we need headsets.
What would you advise ?
Thanks
Thibault
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
Thibault Lamy wrote:
Do anyone have any experience with SIP phone that support
a headset ? We have Budgetone phones but we need headsets.
I am using Snom 190's with Snom head sets and like them a lot.
On my list of things to do is using the Snom with a Labtec PC headset
and see if that works as
Dennis Webb wrote:
I tried getting this to work and gave up when the polycom admin guide
said the park button didn't work in SIP or MGCP applications. Don't
know why it's there.
If you look in the archives you will see a sip debug that I posted
from when you press park on a IP600. The button
Grandstream are supposed to be releasing a BT103 ? Its a 100 series
phone with headphone jack... when, I couldn't say though.
Thibault Lamy wrote:
Hi there,
Do anyone have any experience with SIP phone that support
a headset ? We have Budgetone phones but we need headsets.
What would you advise
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of izo
Sent: Thursday, February 24, 2005 5:12 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] High capacity voicemail
Hi,
Does anybody has experience with high capacity PSTN
On Thu, February 24, 2005 11:59 am, Thibault Lamy said:
Do anyone have any experience with SIP phone that support
a headset ? We have Budgetone phones but we need headsets.
Just deployed a batch of Sipura SPA-841's. Headset jack is standard so a
few of us are using our cell-phone headsets and
Greetings,
First, you are hereby admonished for asking programming/development
questions on the user list.
C does not have a way to do this directly. Yes, you could use some
preprocessing macro but the code would be a nightmare.
You actually gave yourself the answer in your questions.
You can
There was some discussion about this on the list a few months ago. It seems
like there was an issue with inode directory space for that many files or
not enough descriptors. I can't recall exactly, but you will want to do
more homework to make sure the filesystem can handle the quantity of files
The Uniden UIP200 is a decent phone with a headphone jack if the Sipura
doesn't appeal to you.
-Original Message-
From: Thibault Lamy [mailto:[EMAIL PROTECTED]
Sent: Thursday, February 24, 2005 11:00 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] SIP Phone with
1 - 100 of 221 matches
Mail list logo
