Hi Varun -
What are the basic packages required to have
a basic asterisk PBX up and running with all functionality.
I am using fedora 3.
I have downloaded asterisk-1.0.3.
Do I need any other package ?
You'll also need zaptel and libpri. You can download these from the
Asterisk site (http://
I'm trying to get the X100P card working in AU.
So far I have managed to get it to handle incoming calls from the PSTN
and have managed to eliminate pretty much most of the echo.
My big problem is getting the outbound calls to work. When I get ZAP to
dial out it won't connect and I get what I th
Does the call completion feature on the Snom phones
work with Asterisk? If yes, how?
Regards
Thorben
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update o
> I have placed a fxsks=1-8 in my /etc/zaptel.conf . In my
> /etc/asterisk/Zapata.conf if I try to change my channel => 1-4 to
> channel=>1-8 I get errors that it cannot init channel #5.
Have you tried testing the card by pulling the "old" one out and
leaving the new one in, then leaving the conf
Quoting Jon Bebeau <[EMAIL PROTECTED]>:
> HI all - I know, slightly off list, but.. I'm looking for a NPA NXX database
> with City and State.
The North American Numbering Plan Admistrator has some info at
http://nanpa.com/nas/public/assigned_code_query_step1.do?method=resetCodeQueryModel
You
Nevermind, I got it…I had my
sip.conf file dtmf setting wrong.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of David Ishmael
Sent: Saturday, December 18, 2004
1:03 AM
To: 'Asterisk
Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users]
Grandstr
How should my outgoning spool file look like in order to call
using the sip channel (in this example using the Nikotel account)
I tried this, but this is not working.
Channel: Sip/[EMAIL PROTECTED]
Callerid: 1
MaxRetries: 1
RetryTime: 60
WaitTime: 30
Context: common
Extension: 500
Priority: 1
500
I finally
got my Asterisk all setup and everything seems to be working except for menu
interaction between my Grandstream Budge Tone 100 and my Asterisk. I have the
SIP phone setup to properly connect when pressing the ‘Message’
button and that’s working perfectly. When the menu starts, i
On Fri, 17 Dec 2004 20:35:49 -0500, Jon Bebeau <[EMAIL PROTECTED]> wrote:
> HI all - I know, slightly off list, but.. I'm looking for a NPA NXX database
> with City and State. Actually it's for an Asterisk routing app I'm working
> on. I see several vendors that want a few bucks to those that wa
> On Fri, 17 Dec 2004, Jon Bebeau wrote:
>
> > HI all - I know, slightly off list, but.. I'm looking for a NPA NXX
> > database with City and State. Actually it's for an Asterisk routing app
> > I'm working on. I see several vendors that want a few bucks to those
> > that want an arm and leg. I
Hello,
I am using fedora 3.
We have a 2 telephone lines in our office.
I want to setup a asterisk PBX to support say
8 to 10 phones lines.
What is the basic hardware required ?
Thanks
Varun
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
ht
Hello,
What are the basic packages required to have
a basic asterisk PBX up and running with all functionality.
I am using fedora 3.
I have downloaded asterisk-1.0.3.
Do I need any other package ?
Thanks
Varun
___
Asterisk-Users mailing list
[
On Fri, 17 Dec 2004, Jon Bebeau wrote:
> HI all - I know, slightly off list, but.. I'm looking for a NPA NXX
> database with City and State. Actually it's for an Asterisk routing app
> I'm working on. I see several vendors that want a few bucks to those
> that want an arm and leg. I expect this
Mike Derouin wrote:
I have found the 'local calling guide' very helpful - it is free, and not
guaranteed to be accurate - but very good. It has NPA NXX, Rate Centers,
switches, LATA lookups, though I am not sure if they would provide the
source database.
http://members.dandy.net/~czg/search.html
I have found the 'local calling guide' very helpful - it is free, and not
guaranteed to be accurate - but very good. It has NPA NXX, Rate Centers,
switches, LATA lookups, though I am not sure if they would provide the
source database.
http://members.dandy.net/~czg/search.html
Mike Derouin
Ab
I've just created a SJPhone page with screen shots for those new to
asterisk or anyone trying to get the more current version of SJPhone
working. Feel free to send me any feedback directly.
http://www.jimradford.com/asterisk/sjphone/
Regards,
Jim
On Fri, 17 Dec 2004, [EMAIL PROTECTED] wrote
Andrew Kohlsmith wrote:
On December 15, 2004 10:50 pm, Eric Bishop wrote:
This is not fantastic tech support from Digium!
You need to remember that mantis is "run" by a lot more people than Digium
employs. The bug marshalls were likely just holding to the credo that the
bug tracker is for *aste
I was wondering if there are any settings in Asterisk and/or in SIP
clients such as the Sipuras, which will optimize the connections for
DTMF rather than voice?
For Sipuras set -> sip.conf to dtmfmode=rfc2833
And in the Sipura config -> DTMF_Tx_Method[1] "AVT" ;
It works perfect
Jon,
contact me offlist and I can point you to a source
that is very inexpensive.
I also do ss7 consulting and have to have access to
this type of information.
Tom Chandler
[EMAIL PROTECTED]
- Original Message -
From:
Jon Bebeau
To: [EMAIL PROTECTED]
Sent: Friday
Hi,
Can anyone help? I get the following error when trying to complie the h323
channel under the source installation directory
asterisk/channels/h323
i have read the readme file and kept to the recomended versions; h.323 v1.12.2
and PWLIB v1.5.2
Thanks in advance
[EMAIL PROTECTED] h323]#
Hi folks
I again built asterisk cvs with openh323 and pwlib janus as well
as chan_oh323, but this time on Debian Sarge since my passive
AVM Fritz card capi driver wouldn't work on FC3.
Anyway, while my original FC3 build seemed to work great, as far
as I can tell since I just did some initial ste
Thank you!!
--
Christopher Dobbs
Brian Wilkins wrote:
All,
Enjoy these free prompts as an addition to your sounds collection. I hope
you find them useful. You can find them attached to this message.
___
Asterisk-Users mailing list
All,
Enjoy these free prompts as an addition to your sounds collection. I hope
you find them useful. You can find them attached to this message.
account-balance-is.gsm
Description: Binary data
lunch.gsm
Description: Binary data
to-hear-your-account-balance.gsm
Description: Binary dat
That is correct, and the last time I checked, they sell subscriptions for
a monthly charge (depending on frequency of updates) or a one-time charge
of $750 for a single copy.
Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 236-5815
On Fri, 17 Dec 2004, Dave DeChellis wrote:
>
Jon Bebeau wrote:
HI all - I know, slightly off list, but.. I'm looking for a NPA NXX
database with City and State. Actually it's for an Asterisk routing
app I'm working on. I see several vendors that want a few bucks to
those that want an arm and leg. I expect this is published somewhere
by
HI all - I know, slightly off list, but.. I'm
looking for a NPA NXX database with City and State. Actually it's for an
Asterisk routing app I'm working on. I see several vendors that want a few
bucks to those that want an arm and leg. I expect this is published
somewhere by some governmen
On Fri, 17 Dec 2004, Brent Goran wrote:
> We have an application which is primarily DTMF driven (automated on both
> sides), which we are trying to deploy over VOIP and Asterisk (using some
> Sipuras and some IAXY's).
>
> We are finding that in around half the cases, the Asterisk server can't
> de
-Original Message-
>From: Darren Wiebe [mailto:[EMAIL PROTECTED]
>Sent: Friday, December 17, 2004 6:55 PM
>To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
Discussion
>Subject: Re: [Asterisk-Users] ASTCC in production
>
>
>No there is not. That would probably be easy eno
We have an application which is primarily DTMF driven (automated on both
sides), which we are trying to deploy over VOIP and Asterisk (using some
Sipuras and some IAXY's).
We are finding that in around half the cases, the Asterisk server can't
decode the DTMF digits from the field office (or at le
On Fri, 17 Dec 2004, Patrick Campbell wrote:
> Come to think of it since the DTA310 uses DNS to find the SIP server,
> you could setup a DNS cache and override the DNS entry for what packet8
> uses (proxy2-eqix-sjo.packet8.net : 15062, over here) to point to the IP
> of your own SIP server? Kind
> In part I'm trying to figure out at extactly what point it
> might make sense for a small office to consider a T-1 vs DSL
> incombination with POTS or BRI. Also, I'm just very curious.
>
> Michael
> --
> Michael Graves [EMAIL PROTECTED]
> Sr. Product Specialist
On Fri, 17 Dec 2004 17:13:34 -0700, Damon Estep wrote:
>A T1 is made up of 24 64kbps channels (actually they are just timing
>slots on a single channel). When used for data you can use all channels
>or just a few, the CSU/DSU is the device that is configured for what
>channels are used.
>
>In voic
On Dec 17, 2004, at 5:35 PM, Kristian Kielhofner wrote:
Hello,
Can anyone out there confirm as in "Yes I am doing this right now"
that this can be done? I know that the stuff from 2.6 was backported
to 2.4.26 - and it works there (so says the wiki) but before I buy a
bunch of hardware (or don'
> Mark Farver wrote:
>
> > On Fri, 2004-12-17 at 16:26 -0600, Kristian Kielhofner wrote:
> >
> >>We are getting pricing and one provider is telling us
> that they have
> >>quality issues with the "Integrated Access" product. From
> what they
> >>say it sounds like you can have audio drop
Title: Message
I am
pressing the HOLD button on the GS phone
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Christopher DobbsSent: Friday, December 17, 2004 5:03
PMTo: Asterisk Users Mailing List - Non-Commercial
DiscussionSubject:
[EMAIL PROTECTED] wrote:
> Sorry for the misspelling... Thanks for the replies. I will
> set it up and start playing. This is all very exciting.
> I've been using VoIP as my primary phone but this is going a
> bit further. At the office we have a T1 that is probably
> fairly dead after hours.
A T1 is made up of 24 64kbps channels (actually they are just timing
slots on a single channel). When used for data you can use all channels
or just a few, the CSU/DSU is the device that is configured for what
channels are used.
In voice you can use CAS t1, where each 64 channel has inband signali
RTP re-invite is possible. The mess that is Cisco CallManager
supports SCCP, SIP, H.323 and MGCP. Calls from anyone of those
technologies to any other technology works with the signalling
passed through the CCM server and RTP re-invites occur between
the endpoints. So CDR works and scalability d
No there is not. That would probably be easy enough to code in but I'm
not aware of it being done yet.
Darren Wiebe
[EMAIL PROTECTED]
Karl H. Putz wrote:
I am looking for the most stable version of Asterisk to use with ASTCC
for a production environment.
It does not appear that any of the Sta
OK. Now I show my ignorance.
What's the difference between a T-1 and a channelised T-1? I see that
Covad's voip service (formerly GoBeam) requires a channelised T-1. Then
I read recently on the list that many T-1s being installed are actually
HDSLwhich would be not a T-1 at all...right?
Micha
Michael Graves wrote:
On Fri, 17 Dec 2004 17:54:11 -0500, Nabeel Jafferali wrote:
I recently bought a bunch of IP500s and before shipping / tax
they were $170 / each (including power supply). We are lucky
to have received such a great discount, but there's no reason
to pay more than $200 for an IP
On Fri, 17 Dec 2004 17:54:11 -0500, Nabeel Jafferali wrote:
>> I recently bought a bunch of IP500s and before shipping / tax
>> they were $170 / each (including power supply). We are lucky
>> to have received such a great discount, but there's no reason
>> to pay more than $200 for an IP500.
>
>Ho
Mark Farver wrote:
On Fri, 2004-12-17 at 16:26 -0600, Kristian Kielhofner wrote:
We are getting pricing and one provider is telling us that they have
quality issues with the "Integrated Access" product. From what they say
it sounds like you can have audio dropouts on the voice channels when
th
Jeremy,
I am having similar problems, cannot get the phone to register.
I see your config, is there anything special you have done to get the phone
to register? I get the #3 REGISTER ERROR on the lcd?
Thanks,
Chuck
-Original
Message-From: "Jeremy Gehris" <[EMAIL PROTECTED]>To:
Its working now, I have no clue as to what I did, thanks
anyways for being there :)
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jeremy
GehrisSent: Friday, December 17, 2004 6:09 PMTo:
[EMAIL PROTECTED]Subject: [Asterisk-Users] Call Queue
Uniden UIP 200 not working
I j
On Friday 17 December 2004 23:04, Patrick Campbell wrote:
> Come to think of it since the DTA310 uses DNS to find the SIP server, you
> could setup a DNS cache and override the DNS entry for what packet8 uses
> (proxy2-eqix-sjo.packet8.net : 15062, over here) to point to the IP of your
> own SIP s
On Fri, 2004-12-17 at 16:26 -0600, Kristian Kielhofner wrote:
> We are getting pricing and one provider is telling us that they have
> quality issues with the "Integrated Access" product. From what they say
> it sounds like you can have audio dropouts on the voice channels when
> the data
We just received and installed the second TDM card for our
asterisk box. It is installed and gets all the green lights. As well my Debian
box lists the modules as found.
I have placed a fxsks=1-8 in my /etc/zaptel.conf . In my
/etc/asterisk/Zapata.conf if I try to change my channel => 1-4
Nabeel Jafferali wrote:
I recently bought a bunch of IP500s and before shipping / tax
they were $170 / each (including power supply). We are lucky
to have received such a great discount, but there's no reason
to pay more than $200 for an IP500.
How do the Polycom IP500/600 phones compare to th
I just got my wife
this phone, and when calls come in on the queue it will not ring, if I dial the
ext it rings, here is the config from the tftp server:
Thanks
unidencom.txt
# UIP200 Mass
Configuration System Generic File# Notes:# 1. Lines start with '#' are
comments# 2. To leave a fi
Throw that troll phone crap away and write a better one in PHP using AGI. We
did and it works great. We store all NPANXXs into a database with the rates
from multiple carriers. When a call comes in, the PHP script first
determines which state the call came from, and what state the call is going
to.
Come to think of it since the DTA310 uses DNS to find the SIP server, you
could setup a DNS cache and override the DNS entry for what packet8 uses
(proxy2-eqix-sjo.packet8.net : 15062, over here) to point to the IP of your
own SIP server? Kind of a hack but it should work as long as it's running
o
Hello,
> I don't have a great grasp as to what Asterick is capable of, but my
> thoughts were that perhaps with VoIP telephone lines (either hooked up to
> the company's network or just using a 3rd party VoIP provider such as
> Packet8, which is whatI have for personal use) and an Asterick server,
> I recently bought a bunch of IP500s and before shipping / tax
> they were $170 / each (including power supply). We are lucky
> to have received such a great discount, but there's no reason
> to pay more than $200 for an IP500.
How do the Polycom IP500/600 phones compare to the Cisco 7940/7960
ph
Thanks for that. I just got rid of packet 8 and went with 100% asterisk
in my house.
But I use the PAP2-NA and RT31P2 from Linksys for my FXS ports. But
would
like to have an extra FXS laying around just in case..
.o---o.
Brian Fertig
Net
On Fri, 2004-12-17 at 12:01 -0700, Nihal wrote:
> I've freshly installed Asterisk on a Fedora C2 machine. Dual P4's, 2GB RAM.
> 15KRPM Drive.
> Using the default configs and added one Soft Sip phone.
>
> While listening to the demo the quality isnt very good. It's kind of crackly
> and skips a b
> My packet8 "dta310" adapter has the SIP server hardcoded into
> it. If I could change that, I could use that?
Search on broadbandreports.com VoIP forum - there are several postings
(including a few by me) with instructions on how to downgrade the
DTA-310 to v, put in the SIP settings and upg
Hello,
Can anyone out there confirm as in "Yes I am doing this right now" that
this can be done? I know that the stuff from 2.6 was backported to
2.4.26 - and it works there (so says the wiki) but before I buy a bunch
of hardware (or don't buy hardware, depending on how you look at it) I
woul
Hello,
I am currently pricing out various T1 and PRI options for a client of
mine. We need voice and data - we want T's. Whether it be two seperate
T's, two superate fractional T's, or one combined fractional T, we need
it done.
We are getting pricing and one provider is telling us that the
>What codec is your soft phone using?
>Some of the codecs stink, also is the link to the * server heavily used?
I'm using the X-Lite soft phone, it has these codecs selected G711u G711a GSM,
iLBC SPX. I'm not sure which one it ends up using though.
I played with turning off all but GSM, and it d
Sorry for the misspelling... Thanks for the replies. I will set it up and
start playing. This is all very exciting. I've been using VoIP as my
primary phone but this is going a bit further. At the office we have a T1
that is probably fairly dead after hours. Supporting 5-10 users should be
fi
thanks
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Shoval Tomer
Sent: Friday, December 17, 2004 4:43 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Call on hold disconnects...
Also, you can always park th
[EMAIL PROTECTED] wrote:
> I am looking to help out my company find a more budget
> conscious but reliable way to hold conference calls between
> 5+ people. 4x a month we hold several hour long conference
> calls during non-business hours. All of the employees have
> high speed internet. Current
What codec is your soft phone using?
Some of the codecs stink, also is the link to the * server heavily used?
--
Christopher Dobbs
Nihal wrote:
Does some hardware just not work very well with Asterisk?
I've got a fresh installation on a Fedora C2, P4x2, 2GB Ram.
While listening to the demo over a s
On Friday 17 December 2004 21:42, Nihal wrote:
> Does some hardware just not work very well with Asterisk?
Yes. (or, no, depending on how you view the question)
> I've got a fresh installation on a Fedora C2, P4x2, 2GB Ram.
Some people have reported problems with FC3, I don't know if FC2
Are you having the phone place the person on hold, or are you having *
place them on hold?
I dial #700 and it puts them on hold and they stay there,
it also reads off to me the number I dial to get them off hold.
REF: /etc/asterisk/features.conf
--
Christopher Dobbs
Shoval Tomer wrote:
Th
The way I have it set up, is that the mailbox is the same as the exten.
I then wrote a macro that does it for me.
[macro-stdiax]
; ARG1 = User
; ARG2 = Voice Mail Number
exten => s,1,Dial(IAX2/${ARG1}/[EMAIL PROTECTED]||Ttr)
;exten => s,2,Voicemail(u${ARG2})
;exten => s,3,Hangup
;exten => s,10
Does some hardware just not work very well with Asterisk?
I've got a fresh installation on a Fedora C2, P4x2, 2GB Ram.
While listening to the demo over a softphone (over the LAN) I get a number of
crackles and skips.
IS THIS NORMAL FOR ASTERISK?
Or is it hardware related?
Thanks,
Nihal
___
Also, you can always park the call instead of holding it.
> -Original Message-
> From: Ferguson, Michael [mailto:[EMAIL PROTECTED]
> Sent: Friday, December 17, 2004 11:11 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [Asterisk-Users] Call on hold disconnec
Then replace ${CALLERIDNUM} with your
extension/voicemailbox and it will let you in.
No security, but sounds like you don’t
want it.
Tim
From:
[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
On Behalf Of Ross Kevlin
Sent: Friday, December 17, 2004
1:22 PM
To:
[EMAIL
That's both true and false.
We have a legacy PBX here. Panasonic make.
Analog extensions connected to it (a.k.a "stupid" extensions) behace exactly
like the grandstream - you can put a call on hold, but if you put the handset
back on the cradle it's bye bye Mary.
Digital extensions (a.k.a "smar
On Friday 17 December 2004 21:25, Ross Kevlin wrote:
> this would still only work if the mailbox number was the same as the caller
> id. I need some way to get the actual mailbox number of the caller.
Where / how are your mailbox numbers stored?
It shouldn't be too difficult to create a script o
On Friday 17 December 2004 21:10, Ferguson, Michael wrote:
> Antony,
> Thanks. It seems that the GS will not keep the call on hold.
> In the real world though, when you place a call on hold, it is held until
> further action.
Yes, although I might think that hanging up is a further action?
> The
So for newby users of SJPhone... can you tell us exactly what goes in
what box to connect to a standard AsteriskPBX using the latest
interface. I've had no luck so far.
thanks...
On Wed, 2004-12-08 at 09:40, Girish Gopinath wrote:
> Hi,
>
> --- Norman Zhang <[EMAIL PROTECTED]> wrote:
> > I'm fol
> I'm guessing this is a problem with the phone itself. We all have Cisco
> 7960s with 7.3 firmware.
>
> I can yell down the hall at 3091 and say "are you on the phone?" and she
> yells back "no".
> But when I try and call her extension, I get this:
>
> -- Executing Dial("SIP/3044-8eb6", "SIP/309
Greetings…
I’ve been playing with the TrollPhone Rate Engine Addon for a
week or so. I’m curious what is being used out there for LCR
applications?
I’ve run into a stump with the Rate Engine and that is the
costing is done with an integer. With this, how do you put in say 0.014
On Friday 17 December 2004 21:00, Patrick Campbell wrote:
> I am looking to help out my company find a more budget conscious but
> reliable way to hold conference calls between 5+ people. 4x a month we
> hold several hour long conference calls during non-business hours. All of
> the employees ha
Patrick hi.
Asterisk can do that, and you don't need VOIP lines.
If you connect Asterisk to the net, and all employees have a VOIP phone
(either hardware or software) then you're good to go.
What do you need?
To begin with, install linux on an old pc (well, not too old).
Then go to voip-info.org a
this would still only work if the mailbox number was the same as the caller
id. I need someway to get the actual mailbox number of the caller.
- Original Message -
From: Keith O'Brien
To: [EMAIL PROTECTED]
Sent: Friday, December 17, 2004 3:48 PM
Subject: Re: [Asterisk-Users] voicemail w
> In your extensions.conf create a hint:
>
> exten => 215,hint,SIP/215
>
> On the snom phone(s) subscribe the button to:
> destination:
>
> Where 192,168.0.200 is the ip of your asterisk server.
>
> When extension 215 is called, the light on the subscribed button on
the
> snom phones is light
On Fri, 2004-12-17 at 05:43 -0800, Steve Edwards wrote:
> A lot of people are going for the "VOIP only" approach, but SBC says you
> have to have an active analog voice circuit before they will sell you DSL.
>
> Does anybody know which DSL providers will sell you DSL without making you
> pay for
> > > So given all that, I'm looking for ideas and solutions that others have
> > > implemented to address this issue.
> >
> > There are at least two solutions available:
> >
> > 1.) Locate the emergency number for your local 911 provider - every
> > single 911 office should have a non-911 number t
On Fri, 17 Dec 2004, Patrick Campbell wrote:
> I don't have a great grasp as to what Asterick is capable of, but my
> thoughts were that perhaps with VoIP telephone lines (either hooked up
> to the company's network or just using a 3rd party VoIP provider such as
> Packet8, which is whatI have for
On Fri, 17 Dec 2004, Christopher L. Wade wrote:
> Ed Robbins wrote:
> > So given all that, I'm looking for ideas and solutions that others have
> > implemented to address this issue.
>
> There are at least two solutions available:
>
> 1.) Locate the emergency number for your local 911 provider -
You need to compile the conf file for 7905's with the cisco compiler
utility.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matthew
Marlowe
Sent: Friday, December 17, 2004 6:09 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Cisco 7905g TFTP Configur
On Fri, 17 Dec 2004, Michael Graves wrote:
> On Fri, 17 Dec 2004 09:42:28 -0600 (CST), Joe Greco wrote:
>
> >> www.Covad.com
> >>
> >> I have their TeleSoho dedicated loop DSL. It costs the same as the
> >> bundled loop.
> >
> >ADSL or SDSL? (I haven't looked at Covad's pricey offerings for a whi
Antony,
Thanks. It seems that the GS will not keep the call on hold.
In the real world though, when you place a call on hold, it is held until
further action.
The caller will hear messages, music, anything while you are gone to look for a
file, etc.
Technically, if you place the call on hold and
So, the GS is out.
Any recommendations for a Polycom dealer?
Thanks
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ferguson,
Michael
Sent: Friday, December 17, 2004 3:52 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asteris
I'm interested in working on this project, please contact me if you
plan or are actually working on this and we can probably coordinate
something here.
Regards,
Shidan
shidan at gmail
On Fri, 17 Dec 2004 09:05:13 -0500, Noah Miller <[EMAIL PROTECTED]> wrote:
> > I wonder if there is an applicati
I am looking to help out my company find a more budget conscious but
reliable way to hold conference calls between 5+ people. 4x a month we hold
several hour long conference calls during non-business hours. All of the
employees have high speed internet. Currently we dial up an AT&T conf using
re
we also have observed that you must reboot the snom phones EVERY time
you reload the dial plan or restart the server we are using *Asterisk
1.0.0
On Fri, 2004-12-17 at 21:47 +0100, Joris Trooster / Interstroom wrote:
> In your extensions.conf create a hint:
>
> exten => 215,hint,SIP/215
>
> On t
On Friday 17 December 2004 20:43, Ferguson, Michael wrote:
> OK. I guess I was not clear. Sorry.
>
> The phone rings.
> The person picks up the handset and speaks to the caller.
> He then puts the call on hold by pressing the "HOLD" button on the GS
> 100 phone.
> The caller hears music on hold.
Nabeel,
Thanks.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Nabeel
Jafferali
Sent: Friday, December 17, 2004 3:47 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Call on hold disconnects...
[EMAIL PROTECTE
> A lot of people are going for the "VOIP only" approach, but SBC says you
> have to have an active analog voice circuit before they will sell you DSL.
>
> Does anybody know which DSL providers will sell you DSL without making you
> pay for a voice circuit?
I'd say their words are suggesting yo
I did something like this
with some IF logic and changed the callerid to the appropriate callerid for the
mailbox number. Granted I am sure that there is a more eloquent
approach to this but it works for me.
In your extensions.conf create a hint:
exten => 215,hint,SIP/215
On the snom phone(s) subscribe the button to:
destination:
Where 192,168.0.200 is the ip of your asterisk server.
When extension 215 is called, the light on the subscribed button on the
snom phones is light up.
Regards,
Joris
Nethe
[EMAIL PROTECTED] wrote:
> The caller hears music on hold.
> The hand set is placed back on the cradle (as is done on a
> regular phone with a hold button) The call is disconnected.
>
> Is this normal on a IP phone? I think not.
> Does this mean that the GS100 does not really place the call on hol
OK. I guess I was not clear. Sorry.
The phone rings.
The person picks up the handset and speaks to the caller.
He then puts the call on hold by pressing the "HOLD" button on the GS
100 phone.
The caller hears music on hold.
The hand set is placed back on the cradle (as is done on a regular phone
w
On Friday 17 December 2004 20:24, Ferguson, Michael wrote:
> G'Day All,
>
> How do I fix this:
>
> I receive a call at the extension. Press the hold button. Music on hold
> starts. When I place the handset back on the cradle, the call gets hung
> up/disconnected. The Phone is A GrandStream Budge T
hi,
> I receive a call at the extension. Press the hold button. Music on hold
> starts. When I place the handset back on the cradle, the call gets hung
> up/disconnected. The Phone is A GrandStream Budge Tone 100.
this seems a phone problem.
2 solutions:
* don't put the handset back on the cradle
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