Here's a real simple question for the Asterisk Venerable and Wise Ones:
Help me understand how to name my section for the
SIP user, so there is any hope of it ever being used in my sip.conf
file.
The Wiki says that it tries to match the user name from the From: header
in the INVITE packet. If
I've used Kannel (www.kannel.org) for long time; it works out Asterisk,
but it's a good solution sometimes
Use de latest CVS version!
Best regards
hi
list,
does
anyone know how to configure asterisk to be able sending
and
receiving SMS over my own SMS gateway?
it
is connecte
>>context=capi-in
>>devices=2
>
>
> This is just one section which two sets of options. You need to define two
> sections with [...]. See README.
>
> Armin
>
>
no in or out call if i do that (with or without [interfaces]):
[general]
nationalprefix=0
internationalprefix=00
rxgain=0.8
txgain
On Wed, 14 Dec 2005, stéphane plichon wrote:
> >>context=capi-in
> >>devices=2
> >
> >
> > This is just one section which two sets of options. You need to define two
> > sections with [...]. See README.
> >
> > Armin
> >
> >
>
> no in or out call if i do that (with or without [interfaces]):
Quick question,
I have an AVM C4 connected to a "Mehrgeräteanschluss". What should I put
into the /etc/isdn/capi.conf?
Putting P2P works, but I think is wrong. P2MP does not work (CAPI
modules load, but capiinfo says no CAPI installed).
Any help is greatly appreciated.
--
Best regards
Peer
Hi all,
My imaginary scenario is the following one: I have a PSTN gateway called
TotalControl1000, and I want to know, if connecting it to a visible
server with asterisk with public IP I could config it to offer customers
services of virtual switchboard. The customer would save the cost of a
telep
Armin Schindler wrote:
> On Wed, 14 Dec 2005, stéphane plichon wrote:
>
context=capi-in
devices=2
>>>
>>>
>>>This is just one section which two sets of options. You need to define two
>>>sections with [...]. See README.
>>>
>>>Armin
>>>
>>>
>>
>>no in or out call if i do that (with or wit
Hi,
I also experienced broken page receiving fax on asterisk + spandsp
with Digium TE410P. I also tried diff. versions of spandsp and
asterisk, still no luck.
I had no issues using the same asterisk + spandsp config with TE110P.
Any ideas?
At 09:21 AM 11/24/2005, you wrote:
Hi, All
Hi all,
when calling to a queue that has no agents logged in we expect to hang up, here
is the
extensions.conf queue configuration.
exten=> 2020,1,Answer
exten=> 2020,2,Ringing
exten=> 2020,3,Wait(2)
exten=> 2020,4,Queue(gestoria)
exten=> 2020,5,Hangup
But althougth there isn't any agent it let
On Wed, 14 Dec 2005, stéphane plichon wrote:
> Armin Schindler wrote:
> > On Wed, 14 Dec 2005, stéphane plichon wrote:
> >
> context=capi-in
> devices=2
> >>>
> >>>
> >>>This is just one section which two sets of options. You need to define two
> >>>sections with [...]. See README.
> >>>
On Wed, 14 Dec 2005, Peer Oliver Schmidt wrote:
> Quick question,
>
> I have an AVM C4 connected to a "Mehrgeräteanschluss". What should I put into
> the /etc/isdn/capi.conf?
isdnmode=msn
> Putting P2P works, but I think is wrong. P2MP does not work (CAPI modules
> load, but capiinfo says no CA
TE405p and spandsp works good.
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In article <[EMAIL PROTECTED]>, [EMAIL PROTECTED]
says...
> Every few days our receptionist's phone will not take calls on one of
> the extensions. We have an extension 118 going to the first two lines of
> her phone and extension 101 going to the other. If we try to dial 118 it
> goes to voice
Hi all,
we have the need for alot of plain analog lines. We thinking of buying
the new Wildcard TDM2400P. Does anybody have any comments with using
this card, with any version of Asterisk, (maybe ill make this one
Asterisk 1.2.x). I have had some stabilty issues using the 4 TDM400P.
What a
GK> How did you install mpg123? If you installed it with the package
GK> management system, then use the package management system on your
GK> OS to remove it. If you installed it manually, you'll need to remove
GK> it manually.
GK> To actually allow format_mp3 to work you also need to change
G
Can you define a LOT of pots line?
Have you considered a channel bank. Here I'm running an ADTRAN 750. It's
painless. You just need 1 T1 interface card for 24 lines.
Jacques
yusuf wrote:
Hi all,
we have the need for alot of plain analog lines. We thinking of
buying the new Wildcard TDM2400
Hello Armin,
thanks for the quick response.
I have an AVM C4 connected to a "Mehrgeräteanschluss". What should I put into
the /etc/isdn/capi.conf?
isdnmode=msn
isdnmode=msn is in /etc/asterisk/capi.conf, but what about the
/etc/isdn/capi.conf <--- the configuration file for the capi module
HI I am new to asterisk. Could anyone pls tell me how do i create voicemail boxes Best Regards Vinod
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Hey,
When users call my phone at the office, I also want
my mobile to ring..
This works fine using dial(...&...), but there
may be some problems with the cdr's generated from this.
There is only one cdr generated (for the first
destination). I need to see if the call is answered by the
After installing Asterisk, the first thing you'll
need to do is add an extension.
In the process of adding the extension, you can
activate whether you want that extension to have voicemail or not.
Have Fun
Dakota
- Original Message -
From:
Vinod
To: asterisk-users@li
Hi,
try:
joinempty = strict
Morten
- Original Message -
From: "Xavier Gil" <[EMAIL PROTECTED]>
To:
Sent: Wednesday, December 14, 2005 10:19 AM
Subject: [Asterisk-Users] Join when empty problem, in queue
Hi all,
when calling to a queue that has no agents logged in we expect to hang
Hi Guys
After your guies replies now i have changed the machine .But this time i
get little different problem
i made following chnages in sip.conf
[901]
context=fromsip
type=friend
username=901
secret=901
callerid="Test2" <901>
host=dynamic
nat=yes
canreinvite=no
disallow=all
allow=ulaw
dtmfmode=
--- Begin Message ---
Hi Guys
After your guies replies now i have changed the machine .But this time i
get little different problem
i made following chnages in sip.conf
[901]
context=fromsip
type=friend
username=901
secret=901
callerid="Test2" <901>
host=dynamic
nat=yes
canreinvite=no
disallow=a
Hi But that does not create the voice mail boxes. Is there any script which does it as mentioned in this link below http://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html?page=3 Regards VinodDakota <[EMAIL PROTECTED]> wrote: After installing Asterisk, the first thing you'll need to d
On Wed, 14 Dec 2005, Peer Oliver Schmidt wrote:
> Hello Armin,
>
> thanks for the quick response.
>
>
> > > I have an AVM C4 connected to a "Mehrgeräteanschluss". What should I
> > > put into
> > > the /etc/isdn/capi.conf?
> > isdnmode=msn
>
> isdnmode=msn is in /etc/asterisk/capi.conf, but wha
well, guess that's the way it is(a feature? i don't know)...
but you can help ourself with the cmd SetCDRUserField respectively
AppendCDRUserField (see
http://www.voip-info.org/wiki/view/Asterisk+cmd+SetCDRUserField)
for dialing multiple destinations, maybe follow_me could be an interesting
pat
Hi Guys
After your guies replies now i have changed the machine .But this time i
get little different problem
i made following chnages in sip.conf
[901]
context=fromsip
type=friend
username=901
secret=901
callerid="Test2" <901>
host=dynamic
nat=yes
canreinvite=no
disallow=all
allow=ulaw
dtmfmode=
Hi all,
I downloaded version asterisk 1.2.1 on my Debian.
When i did "make", i met this error:
"
build_tools/make_version_h>include/asterisk/version.h.tmp
/bin/sh: line 1: build_tools/make_version_h: permisson non accordee
make: ** [include/asterisk/version.h] erreur 126.
It's not an error of a
Thanks,
This sounds interesting, but it may cause some delay before it rings on my
cell..?
It is possible to implement a "ringall" strategy in the patch?
Morten
- Original Message -
From: <[EMAIL PROTECTED]>
To:
Sent: Wednesday, December 14, 2005 12:31 PM
Subject: Re: [Asterisk-Us
hello,
We installed an asterisk box last weekend dealing with 2 incoming groups
of ISDN lines, and some 15 polycom phones. Works quite OK so far. But we
have some problems that somehow I cannot resolve so far :-(
The most urging issue is getting callerID to work.
When I check logfiles, I see line
Is this normal? Can I ignore this messages?
Dec 14 13:18:06 DEBUG[5312] db.c: Unable to find key
'MGCP/aaln/[EMAIL PROTECTED]' in family 'cfb'
Dec 14 13:18:06 DEBUG[5312] db.c: Unable to find key
'MGCP/aaln/[EMAIL PROTECTED]' in family 'cfb'
[...]
--
Alejandro Vargas
_
___
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France et l'international.
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> Every few days our receptionist's phone will not take calls on one of
> the extensions. We have an extension 118 going to the first two lines of
> her phone and extension 101 going to the other. If we try to dial 118 it
> goes to voicemail even though she is not on the phone. Asterisk is
> t
Hi,
Started getting a bombardment of these messages on the Asterisk console this
morning (20+ a second):
Dec 14 10:00:30 WARNING[17006]: channel.c:588 ast_queue_frame:
Exceptionally long queue length queuing to SIP/bluecity29-a5cfDec 14
10:00:30 WARNING[17006]: channel.c:603 ast_queue_frame: Unab
On 12/14/05, Brian Capouch <[EMAIL PROTECTED]> wrote:
> Douglas Garstang wrote:
>
> >
> > I can't understand why it was implemented this way (lack of design maybe?).
>
> Yep, that's it. Asterisk was designed by a bunch of fools who never
> even gave the first thought to what they were coding up.
>
Rich - Even though I mentioned ethernet failover, I might have made it still
a little too broad. The linux ethernet bonding module has been around for
years, and there are several modes the linux bonding module can use which
include:
mode=0 (balance-rr)
Round-robin policy: Transmit packets in sequ
Hi,
'#' (fast forward) and '*' (Rewind) not working in VoicemailMain with
Asterisk 1.2.1 Do I have to do something in the dialplan to make this
work? I have '##' set as a blind transfer and '*2' set as a attended
transfer in features.conf. Per the Wiki Voicemailmain has the following
settings
I think this is normal if you don't have call-forward-busy enabled. They
key is deleted when it is disabled and added when enabled.
- James
Alejandro Vargas wrote:
Is this normal? Can I ignore this messages?
Dec 14 13:18:06 DEBUG[5312] db.c: Unable to find key
'MGCP/aaln/[EMAIL PROTECTED]' i
Vinod wrote:
Is there any script which does it as mentioned in this link below
http://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html?page=3
A script is no longer needed. Just edit your voicemail.conf file and add
a line for each voicemail box. The first time someone drops into
voicema
Although I do not have an answer I changed the title so maybe someone
with MGCP experience may notice it.
>
> Is this normal? Can I ignore this messages?
>
> Dec 14 13:18:06 DEBUG[5312] db.c: Unable to find key
> 'MGCP/aaln/[EMAIL PROTECTED]' in family 'cfb'
> Dec 14 13:18:06 DEBUG[5312] db.c: U
Hello,
I wish to configure Hylafax in order to send either
fax or voice to Asterisk
I've got a TDM400P (1FXS/1FXO) .
What' s the best way to check the line to send fax or
voice for incoming or outgoing ?
Thanks for help
H.G
> -Original Message-
> From: Vinod [mailto:[EMAIL PROTECTED]
> Sent: Wednesday, December 14, 2005 5:26 AM
> To: asterisk-users@lists.digium.com
> Subject: [Asterisk-Users] voicemail boxes
>
> HI
>
> I am new to asterisk.
> Could anyone pls tell me how do i create voicemail boxes
>
> B
I need help how to config sipura 3000 send and receive calls please.
Thanks-- [EMAIL PROTECTED]
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Hello all,
Please help me with the sipura 3000 how the Asterisk config need send and receive calls from Sipura 3000
What is Asterisk config need to input
Thanks
[EMAIL PROTECTED]
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Hi,
According to http://www.voip-info.org/wiki-Asterisk+video it should be
possible to place video calls using asterisk.
So far I managed to get both Microsoft Messenger and a video conference
system from Tandberg to register with asterisk. Voice calls between both
stations work perfectly (using u
Title: Re: [Asterisk-Users] Hint Priority for Polycom Phones
Hello Doug,
I assume you have subscribecontext set in sip.conf
right?
Also, I have a 601/sidecar and have the hints
working fine on the first registration. On my second server registration
they are not yet coming through.
I am
Hello all.
Has anybody works with appradius? where can I find documentation?
Regards,
Jsalas
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I am trying to register my Asterisk to a Portabilling system. My asterisk
registers with no problems, but when i try to send calls to portabilling I
get the response:
Dec 14 15:23:31 WARNING[420]: chan_sip.c:727 retrans_pkt: Maximum retries
exceed
ed on call [EMAIL PROTECTED] for seqno 104
(N
In article <[EMAIL PROTECTED]>, [EMAIL PROTECTED]
says...
> FC4, Asterisk 1.0.9 and SjPhone softphone. On CLI I get this message
> every 20 sec.
> # Testing 10.0.0.203 with 10.0.0.0
>
> 10.0.0.203 is the IP of softphone and 10.0.0.0 is the network defind in
> sip.conf. Asterisk server is on
>I'd recommend the Digium dual port cards - generation 2 card are
excellent and the support we receive superb.
>And it supports Digiums support and development of Asterisk - Sangomas
contribution is token if any.
Unfortunately I cant speak for the 2nd gen. cards as we haven't used
them. The 1s
> Rich - Even though I mentioned ethernet failover, I might have made it still
> a little too broad. The linux ethernet bonding module has been around for
> years, and there are several modes the linux bonding module can use which
> include:
>
> mode=0 (balance-rr)
> Round-robin policy: Transmit p
> I need help how to config sipura 3000 send and receive calls please.
Go to www.voxilla.com and run their config wizard.
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Hi,
Please excuse the double post but I am about to report this as a bug and
I want to verify that others are having the same problem. Also I have
seen numerous bugs reported that are not bugs but just misconfiguration,
etc. and I do not want to burden the developers with a frivolus bug
repor
\
> >I'd recommend the Digium dual port cards - generation 2 card are
> excellent and the support we receive superb.
> >And it supports Digiums support and development of Asterisk -
Sangomas
> contribution is token if any.
>
> Unfortunately I cant speak for the 2nd gen. cards as we haven't used
Hi!
> Dec 13 12:19:29 WARNING[4112]: loader.c:325 __load_resource:
> libmysqlclient.so.15: cannot open shared object file: No such file or
> directory
Mine is in /usr/lib/libmysqlclient.so, so how about just adding a
symlink?
Philipp
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Hi!
> Is this normal? Can I ignore this messages?
>
> Dec 14 13:18:06 DEBUG[5312] db.c: Unable to find key
> 'MGCP/aaln/[EMAIL PROTECTED]' in family 'cfb'
> Dec 14 13:18:06 DEBUG[5312] db.c: Unable to find key
> 'MGCP/aaln/[EMAIL PROTECTED]' in family 'cfb'
> [...]
Look like you have enabled cal
Hi,
I am planning on restarting asterisk nightly as I seem to be
experiencing some sort of memory leak (Asterisk slows down over time). I
have reviewed the Asterisk suggestions for management and one item is
the routine rebooting of Asterisk. Since I have Asterisk 1.2.1 what is
the recommend
I'm using an Asterisk system with a Zultys MX250 as a media gateway
for our PSTN. When I configured this originally, I couldn't make or
receive any outside calls - the MX250 would actually crash and
restart itself. Very bad. As I was troubleshooting, I found a tip in
the Asterisk wiki that
OOPs I forgot to mention I am using Asterisk 1.2.1 and I had the same
problem with 1.0.9 and 1.2.0
Chuck Bunn wrote:
Hi,
Please excuse the double post but I am about to report this as a bug
and I want to verify that others are having the same problem. Also I
have seen numerous bugs reported
Kris,
I highly recommend the snom 320. Very easy to configure, and very easy
to setup line appearances.
As has already been mentioned, the idea of "lines" is a bit dated.
For more information, read this:
http://forums.digium.com/viewtopic.php?t=891.
Sean
Duracom ISP Lists wrote:
We
Hi
Just checking,
Is there any hardware echo cancellation card available for the digium
TDM400P card
I read the archives and could not find any.
I think I need the TDM2400 card for this
Thanks
Patrick
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Title: Re: [Asterisk-Users] TDM01B vs. X100P
I fixed this by making the extensions on
the asterisk box DIDs …. So when the IPO passed the 3 digit extension number
then the asterisk box looked at that and sent it on to the extension. Asterisk
seemed to see all H323 calls as outside calls in
in queues.conf
change
joinempty = no
leavewhenempty = no
to
joinempty = strict
leavewhenempty = strict
From: Xavier Gil <[EMAIL PROTECTED]>
Reply-To: Asterisk Users Mailing List - Non-Commercial
Discussion
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Join when empty probl
I need a few minutes of time to work out a dial out problem. I'm willing to
pay for your time.
What I have is a system that connect 2 external VMS systems to one of two
Telco T1's. Mainly the Telco T1's route inbound calls to one of the two
external VM systems depending on the DNIS. This part
rossi.tek wrote:
> I'm looking for iax2 wifi phones, do you know where i can buy them?
Yes. Nowhere.
:)
--
Cheers,
Matt Riddell
___
http://www.sineapps.com/news.php (Daily Asterisk News - html)
http://freevoip.gedameurope.com (Free Asterisk Voip Co
http://www.engadget.com/entry/1234000497072377/
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Really? Did you try reading classes?
On 12/14/05, Talkvoip Telecom Canada <[EMAIL PROTECTED]> wrote:
> I need help how to config sipura 3000 send and receive calls please.
> Thanks
> --
> [EMAIL PROTECTED]
> ___
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On 12/14/05, Patrick Fortin <[EMAIL PROTECTED]> wrote:
> Hi
>
> Just checking,
>
> Is there any hardware echo cancellation card available for the digium
> TDM400P card
>
> I read the archives and could not find any.
>
> I think I need the TDM2400 card for this
>
No. Not at this time. You will nee
Patrick Fortin wrote:
Is there any hardware echo cancellation card available for the digium
TDM400P card
No. Software echo cancellation is fine for small density applications
like 4-8 ports, unless you are using a very low performance CPU.
___
--Ba
What I really want to be able todo is use the services button or any of
the other buttons that serve no purpose right now.
I would like to have it start a page (which on my * box is just dialing
a particular extension), I have this working on my Polycom 501's using
the 3rd line appearance, how
Does anyone have an idea of where cisco 7940's get their time from? Up
until monday (when our dns server crapped out so we killed it), our
phones all had time... now they only show time when they're just
rebooted and it's only for a few minutes.
Any ideas?
Aaron Daniel
__
I am looking for IP phones that are headset and then a phone that
clips on your belt. I have been looking at wi-fi phones, but I'm not
sure about headset capabilities. I'm using these with *, so
compatibility with * is required.
Can anyone offer any suggestions?
~kurth
--
Kurth Bemis
___
combine an iaxy and a plain old cordless phone. It works great for me
around the workplace. I think there are even four-line, five-handset
cordless systems, although you'd need four iaxys as well.
Moj
rossi.tek wrote:
I'm looking for iax2 wifi phones, do you know where i can buy them?
Thank
On Wed, 2005-12-14 at 17:33 +0100, Matt Riddell wrote:
> rossi.tek wrote:
> > I'm looking for iax2 wifi phones, do you know where i can buy them?
>
> Yes. Nowhere.
>
> :)
>
not entirely true if you expand your definition :P
I have an ipaq which is capable of acting like a soft phone (and it
do
Hi all,
Has anyone a good piece of advice on using traffic shaping embeded with
*? As in our case it is not possible to configure it in the ADSL router
we would like to implement some kind of bandwidth reservation policy in
*. What about using * with 2 network cards betwen the LAN and ADSL
router
Hi,
I'm using Asterisk 1.2.1, and have been trying to sue the Background()
command followed by Read() (for a screening app, but that's beside the
point)
I did the following
s,1,Background(blablabla)
s,2,Read(VARIABLE||1) ; accepting only one digit (1 to accept call, anything
else to hangup)
...
Dakota wrote:
After installing Asterisk, the first thing you'll need to do is add an
extension.
In the process of adding the extension, you can activate whether you
want that extension to have voicemail or not.
Have Fun
Dakota
Sounds like someone is using [EMAIL PROTECTED] (or at least AM
I'd like to configure Asterisk so that incoming calls from one POTS line
are shared amongst multiple extensions. i.e. If one SIP phone answers
the call, another SIP extension phone can pick up
and join the conversation. How do I configure this in extensions.conf?
If you change your config for your phones to use unicast for your SNTP Mode, the time should stay. I had the same problems until I changed it to unicast.On 12/14/05, Aaron Daniel
<[EMAIL PROTECTED]> wrote:
Does anyone have an idea of where cisco 7940's get their time from? Upuntil monday (when ou
I thougt i have some problems with ztdummy and removed that # in front
of ztdummy in the zaptel Makefile before compiling. But still no change.
I even tried it with another Phone, a Planet VIP-150T. Still the same
Problem, i don´t hear anything from the SIP Phone on the ISDN Phone, but
i hear ev
Kurth:
The UTStarcom F1000 has an ear bud for it. I believe the Hitachi IPF-5000
also has a jack for headset/ ear bud use.
Garrett Smith
<[EMAIL PROTECTED]>
716-250-3408 Direct
716-903-9495 Cell
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kurth Bem
and that ipaq can do iax2 ??
Guess not
cheers
klaus
trixter aka Bret McDanel schrieb:
On Wed, 2005-12-14 at 17:33 +0100, Matt Riddell wrote:
rossi.tek wrote:
I'm looking for iax2 wifi phones, do you know where i can buy them?
Yes. Nowhere.
:)
Hi
I am using asterisk 1.2.1, does any one has any luck with asterisk and h323.
I want to use the codecs 723 and 729 with it.
I am having one way audio issues with oh323 with I receive a call to
asterieks through 723 .
is there a successful implementation ?
regards
kani
___
http://www.krisk.org/astlinux/misc/astshape
kicks butt
hth
-Original
Message-
From: Jose Limeres
[mailto:[EMAIL PROTECTED]
Sent: Wednesday, December 14, 2005
10:21 AM
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: [Asterisk-Users] traffic
shaping
Look at meetme, also FOP (www.asternic.org) can do that for you.
On 12/14/05, Robert La Ferla <[EMAIL PROTECTED]> wrote:
> I'd like to configure Asterisk so that incoming calls from one POTS line
> are shared amongst multiple extensions. i.e. If one SIP phone answers
> the call, another SIP exte
Alessio Focardi wrote:
GK> How did you install mpg123? If you installed it with the package
GK> management system, then use the package management system on your
GK> OS to remove it. If you installed it manually, you'll need to remove
GK> it manually.
GK> To actually allow format_mp3 to work
Let me revise this a little:
I'd like to configure Asterisk so an incoming call from one POTS line is
shared amongst multiple extensions - both SIP and analog. i.e. If one
SIP phone answers the call, another SIP or analog extension phone can
pick up and join the conversation. How do I confi
look into linux advanced routing and traffic control
lartc
Casey Boone
Jose Limeres wrote:
Hi all,
Has anyone a good piece of advice on using traffic shaping embeded with
*? As in our case it is not possible to configure it in the ADSL router
we would like to implement some kind of bandwidth
Jose,
I don't know what everyone else uses, but I use wondershaper. It's a
bit rough, but it does the job well for what I need ( prioritize voip
traffic over everything else ).
Google should bring it up for you.
Sean
Jose Limeres wrote:
Hi all,
Has anyone a good piece of advice on using
Maybe a bit simplistic... ATA with any cordless phone and walmart headset...
Garrett Smith wrote:
>Kurth:
>
>The UTStarcom F1000 has an ear bud for it. I believe the Hitachi IPF-5000
>also has a jack for headset/ ear bud use.
>
>Garrett Smith
><[EMAIL PROTECTED]>
>716-250-3408 Direct
>716-903-949
But can they run Asterisk and create an IAX trunk back to your PBX while
running a softphone?
I thought not.
-Kerry
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of C F
Sent: Wednesday, December 14, 2005 8:44 AM
To: Asterisk Users Mailing List - Non-Com
Cheers all.
I hope I did not miss this in my quick searching for this, and if I did,
apologies, please just a minor scolding as you point me at the URL. But,
if I did not miss it, then is there any one out there who has
figured-out how to skin this cat.
I have a few people's mobile phone numbers
also you can ring multiple extensions:
Dial(SIP/101&SIP/102&SIP/103)
C F wrote:
>Look at meetme, also FOP (www.asternic.org) can do that for you.
>
>On 12/14/05, Robert La Ferla <[EMAIL PROTECTED]> wrote:
>
>
>>I'd like to configure Asterisk so that incoming calls from one POTS line
>>are sh
Sean Cook wrote:
also you can ring multiple extensions:
Dial(SIP/101&SIP/102&SIP/103)
I have that but once one extension picks up, others can't join in.
Well, at least when I tried it with mixed SIP and Zap, it didn't work.
Maybe all SIP does but I need it to work for all phones SIP and
> I did the following
> s,1,Background(blablabla)
> s,2,Read(VARIABLE||1) ; accepting only one digit (1 to accept call, anything
> else to hangup)
That's not the right approach. Do something like his:
[confirmcall]
exten => s,1,Background(blablabla)
exten => 1,1,Goto(accept_call_context,s,1)
exte
a simple m0n0wall or pfsense system running on a sub $100
pc makes a GREAT router for this and allows you to use multiple internet connections in
concurrency for speed increases
other than that, yes, 2 NICs and some creative
networking and you're done
From: [EMAIL PROTECTED]
[mailto:
Hi everyone,
I have started trying to use ChanIsAvail() to detect when a phone is
in use (on any call) and my results are disappointing.
Here are some examples out output to the console followed by the
meaning of the return status code based on what I have found in the
comments on this
I'm trying to compile 1.2.1 for the first time and I am getting a
compile error that I can't figure out. I've compiled 1.0.x many times
without this issue, although it has been on different boxes. The error
is "configure: error: termcap support not found", which is odd because
when I do a "rp
Oops, my bad. 5 minutes after I sent it, I realized I was missing
ncurses-devel.
Peder @ NetworkOblivion wrote:
I'm trying to compile 1.2.1 for the first time and I am getting a
compile error that I can't figure out. I've compiled 1.0.x many times
without this issue, although it has been on
I use this at work. You have to make sure you use the right T / t
options when dialing the mobile, then just use the standard # transfer.
I changed ours to ##.
- James
[EMAIL PROTECTED] wrote:
Cheers all.
I hope I did not miss this in my quick searching for this, and if I did,
apologies, pl
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