Elliot Mackenzie wrote:
I have a situation involving both caller id and distinctive ring in
australia that appears to be having issues. I am using the CVS snapshot
current as of an hour ago.
The distinctive ring was working ok until i arranged for the telco to
turn on callerid: now the distinc
Hi,
I have Siemens combiset - it can gateway GSM phone to normal analog phone.
It has output where I can connect regulat analog phone.
How can I connect to combiset with Asterisk - via fxo or fxs ?
Thanks,
regards,
Robert.
___
Asterisk-Users mailing
I have read that many people have problems with Caller Id in the Uk.
What kind of surprises me is that incoming caller Id workes for me with an
X100P and Asterisk (pre 1.0 release) without any modifications.
I know that in Ireland we use the same wiring and phone connectors as the US
(RJ11 with t
Hi,
I'd like to get some opinions whether is appropriate to run Asterisk in
chroot environment...
I'd also like to hear if anyone is willing to share knowledge or scripts to
make chroot environment for Asterisk ?
Thanks in advance,
Robert.
___
Asteri
Whoops. Apologies for sending this more than once. I thought a
sendmail upgrade had broken, but it was just slow :-)
Someone mentioned there are some patches required to make callerid work
in the UK. Do similar patches apply for use in Australia? How would I
make my own modifications? Is t
> Generally, you follow the directions.
>
> http://www.asteriskpbx.org/index.php?menu=download
Which state:
cvs checkout -r v1-0 zaptel libpri asterisk asterisk-addons asterisk-sounds
So this means you would be getting 1.0? Or?
___
Asterisk-Users mail
I have a situation involving both caller id and distinctive ring in
australia that appears to be having issues. I am using the CVS snapshot
current as of an hour ago.
The distinctive ring was working ok until i arranged for the telco to
turn on callerid: now the distinctive ring detection appe
Doug L. Dawson wrote:
I am attempting to setup a SIP phone that is behind NAT router, to hook
up to my Asterisk server the phone is a Grandstream BudgetTone100 has
anyone had any luck doing this.
add the following to sip.conf for the grandstream's context
host=dynamic
nat=yes
also look at http://ww
Hey guys.
I'm having constant problems with faxing. All my calls come in via PRI
to my t100p, then I have a TDM w/FXS ports that I have the fax machine
plugged into.
About half the time the faxes come in half complete or garbled. I have
echocan off. I'm not sure what else to try.
Anyone have a
I have a question here. If both companies use 200 as their extension, how
can * tell which context a received sip call uses?
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Gene Willingham
> Sent: Monday, November 08, 2004 3:00 PM
>
I am attempting to setup a SIP phone that is behind NAT
router, to hook up to my Asterisk server the phone is a Grandstream
BudgetTone100 has anyone had any luck doing this.
Thanks
Doug D
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Hi group
Anyone has perform Unity SIP integration with Asterisk PBX?
Thanks!
Do you Yahoo!?
Check out the new Yahoo! Front Page. www.yahoo.com___
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To
Is there a change log anywhere that would show changes to * between
current CVS-HEAD and 1.0.2 stable? Not code changes, but feature
changes.
Thanks
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I struggled with this issue too. In my case, it was for the voice part of
a combined T-1 line from XO, which is coming out an Adtran 604 router as
the CPE. The secret sauce for me was to configure all 24 channels, but
then only use the first 6 (since that is what we ordered) in the dialplan.
So, h
On Mon, Nov 08, 2004 at 12:41:40PM -0600, Matthew Boehm arranged a set of bits
into the following:
> Have you tried chan_sccp?
Just a heads-up while schan_sccp doesn't yet support
hold/transfer/voicemail buttons they code for hold and voicemail has
been written it just needs to be tested.
If anyon
nice looking phone and a good price too.
- Original Message -
From: "John Gray" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<[EMAIL PROTECTED]>
Sent: Monday, November 08, 2004 3:08 PM
Subject: [Asterisk-Users] iPeya iPHONE-1001M?
Anybody know anythi
yes
- Original Message -
From:
Uma S. Pandey
To: [EMAIL PROTECTED]
Sent: Monday, November 08, 2004 3:43
PM
Subject: [Asterisk-Users] Same Extensions
in Multiple contexts
Hi
For a customer, I am trying to
setup 3 different companies on one asteri
Charles Osstyn [EMAIL PROTECTED] wrote:
> (Article auto-converted from unnecessary HTML to nice plain text.)
>
>
> Hi all, I am too new with Linux, to really experiment with the callerid.
> I know the problem is due to BT using a different technical platform.
>
> So as too using Perl or any other
Nathan C. Smith wrote:
What would one enter to get the stable or 1.x version of Asterisk and
associated modules via CVS? I've googled and wikkied but I'm using the
wrong terms or asking the wrong questions.
Generally, you follow the directions.
http://www.asteriskpbx.org/index.php?menu=download
_
Hi All,
I use four ITSPs for incomming and (mostly) outgoing calling; NuFone,
VoicePulseConnect, Clearpath and VOIPJet. In their example files some
suggest setting jitterbuffer=yes in IAX.CONF while others say turn it
off. At present I have it turned on.
My incomming calls on VPC DIDs sound reall
All,
I just put FC3 (fedora core 3) on my machine. It uses udev.
I looked at the README.udev in zaptel and put those lines
in the /etc/udev/rules.d/50-udev.rules file.
I rebooted just for grins and I still am getting the message:
Notice: Configuration file is /etc/zaptel.conf
line 147: Unable to op
When you compile asterisk and zaptel, after the make and make install;
try doing a make config. This puts the startup files in
/etc/rc.d/init.d so you can use 'service start zaptel' and 'service
start asterisk' to start, restart, and/or stop the applications. It
takes away the need to start the t
I was able to get this to work. Not sure if it is the best way or the only
way, but this is how I did it. Including contexts do not give you the
desired result. You must keep the contexts separate and use the goto to get
it to work.
Use the internal context in sip.conf. If you don't people ca
Greetings,
Also check that you are really talking to the asterisk server.
We had the same problem this morning with SIPURA when we were talking
x.x.x.100 and not x.x.x.249
The problem was that asterisk listens on the first interface it finds on
the server. So if there are two network cards, or t
asterisk-dev，您好！
x100p drive how to implement detect Disconnect ?
x100p drive detect Tone-based Supervisory Disconnect?
how to implement ?
need to modify zaptel wcfxo.c ?
http://www.cisco.com/warp/public/788/signalling/fxo_disco
On November 8, 2004 06:41 pm, Steve Underwood wrote:
> I have looked at many similar reports, and all but one turned out to be
> due to data slips. However, if you are sure this is only happening with
> one particular FAX machine your problem might be different. In t30.c
> uncomment the first line.
Title: bad quality for toll free calls with gafachi
Hi,
I experienced bad voice quality for toll free calls with gafachi. All other long distance calls are ok. Does anyone have this problem too?
Thanks,
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I had similar problem on FreeBSD with DeadAGI.
In the beginning all worked, but when I configured
noload => pbx_wilcalu.so
not to load CPU, I started to get disconnects with file.c:1058
ast_waitstream_full: Wait failed
Just let asterisk have 99% CPU and load wilcalu and everythong be ok, I
I am experiencing one way audio when I call through Gafachi. I can hear the
person that I am
calling but they can not hear me. I am able to call FWD echo test and have
no problems. My
daughter has called me using sjphone (sip) and firefly (iax2) with no
problems.
I have been using Gafachi since
> Have a new install of RH9 with cvs head from today. Same problem
> with running cvs head from Oct 29th. Calling this box from another *
> box via iax2 results in:
> dial their milliwatt generator, good solid 1k hz tone
> dial an extension with playback, like:
>exten => 6200,1,Playback(de
802.3AF calls for power to be on the data wires
http://standards.ieee.org/getieee802/download/802.3af-2003.pdf
Table 33C.1 and 33C.2 on page 90 clearly states that in mode A
Terminal A is on pin 3
Terminal B is on pin 6
Terminal C is on pin 1
Terminal D is on pin 2
There is a mode B which uses
My email server broke, so I didn't know if this made it to the list the
first time.
Ok I figured it out mostly, I went by what Flynn posted. I commented out
the particular lines in res_features.c when a call is connected no DTMF is
passed. The only problem I am having now is I'm not sure how to
Hi Scott,
I have looked at many similar reports, and all but one turned out to be
due to data slips. However, if you are sure this is only happening with
one particular FAX machine your problem might be different. In t30.c
uncomment the first line. Rebuild spandsp and try FAXing. You should get
Michael Welter wrote:
In my dial plan, I have
exten => 160,1,txfax(${FILENAME})
When I dial 160 from a fax machine, I hear a broken dial tone. The
processor utilization goes to 100%, and the following pair of messages
is displayed at the console:
100% CPU utilisation is not normal, unless you ha
Hello,
On Mon, 8 Nov 2004, Kubat, Philip wrote:
> We currently have an Asterisk installation and need to add cordless /
> wireless phones. Requirements are these phone need to be equals to the
> "wired" devices, i.e. dedicated buttons for hold, transfer, etc. , e.g. not
> an ATA connected analog
>Think of it as a dinky little $0.50 padlock on your storage shed. If a
>thief cuts the lock, they are in a lot more trouble than just opening
>the door.
>
>Separate WLAN (ie not with your normal phones, and not with your
>workstations), and WEP (even 64 bit) will keep people out of it. Not
>havi
On Mon, 2004-11-08 at 16:27 -0600, Michael Giagnocavo wrote:
> >The WiSIP phone supports WEP 128 encryption. Not sure if it supports WPA
> >encryption, but that'd be your best bet. I'd use maximum encryption, and
> >separate your AP from your regular network. Just plug an AP into another
> >Etherne
On Mon, 2004-11-08 at 06:59, Gianni Veloce wrote:
> Dear * Experts,
> I intend to use a laptop for Asterisk at home (because
> of space problems and as I already have one spare).
>
> I would like Asterisk to 'sit between' my ISDN (BRI)
> Line and use my existing ISDN telephone as extension.
>
>
Hello all.
I'm running SpanDSP 0.0.2pre4 with Asterisk v1.0 (10/23 from CVS) and
am having trouble with receiving cut off faxes from a Lexmark 6170 on
the distant end. I'm running this on a Tyan S2420 board that is about
four years old with a P3-800 and 256 MB on it.
I have tested with a PRI conn
In my dial plan, I have
exten => 160,1,txfax(${FILENAME})
When I dial 160 from a fax machine, I hear a broken dial tone. The
processor utilization goes to 100%, and the following pair of messages
is displayed at the console:
Slow carrier up
Slow carrier down
This pair of messages repeats until
Does anyone know how to transfer ADSI information over IAX, I have
looked at the code, and it apears that this is posible.
--
Christopher Dobbs
Software Engineer
Eracew Computer Services
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I have a situation involving both caller id and distinctive ring in
australia that appears to be having issues. I am using the CVS snapshot
current as of an hour ago.
The distinctive ring was working ok until i arranged for the telco to
turn on callerid: now the distinctive ring detection appe
These packages have been compiled on a "bone stock with updates" Fedora
Core 2 installation running a Pentium 4 1.8GHz processor and
kernel-2.6.8-1.521. Check em out:
ftp://ftp.linuxsys.com/pub/releases/FC2/asterisk-v1.0
ftp://ftp.linuxsys.com/pub/releases/FC2/asterisk-CVS
>The WiSIP phone supports WEP 128 encryption. Not sure if it supports WPA
>encryption, but that'd be your best bet. I'd use maximum encryption, and
>separate your AP from your regular network. Just plug an AP into another
>Ethernet card on your Asterisk server. The phones only need to talk to the
Well, the phone automatically does call waiting on each line you
register so you will be able to get a call on each line.
You could always do this
In Asterisk
Setup extensions 100, 101, 102, 103, 104, 105, 106
Set your dial plan to ring 100 on all incoming calls.
Set 100 to roll through 1
From my voicemail.conf,
my context where I define my mailboxes in this file is
[sip]
From: Paul Rodan [mailto:[EMAIL PROTECTED]
Sent: Monday, November 08, 2004 2:49 PMTo: 'Asterisk Users
Mailing List - Non-Commercial Discussion'Subject: RE:
[Asterisk-Users] MWI Doesn't Turn Off
What is
Hello,
I'm about to connect asterisk with Alcatel Enterprise PBX using SIP
trunking, I can't find if Asterisk has this capability. Can you please
advice?
Thank you.
-David
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I am attempting to setup a Polycom SoundPoint 600 in the same manner
that I have a Cisco 7960 (SIP) operate as a receptionist phone. With
the Cisco 7960, I am able to have 6 line appearances all display the
same phone number, and thus give the receptionist the ability to handle
6 simultaneous
Have a new install of RH9 with cvs head from today. Same problem
with running cvs head from Oct 29th. Calling this box from another *
box via iax2 results in:
dial their milliwatt generator, good solid 1k hz tone
dial an extension with playback, like:
exten => 6200,1,Playback(demo-abouttot
What is the [context] you are using in
voicemail.conf ?
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Wiley E. Siler
Sent: Monday, November 08, 2004
3:48 PM
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: RE: [Asterisk-Users] MWI
Doesn't Tur
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Nathan C. Smith
> Sent: Monday, November 08, 2004 12:59 PM
> To: '[EMAIL PROTECTED]'
> Subject: [Asterisk-Users] how to get Stable 1.X via CVS
>
>
>
> What would one enter to get the stable or 1.x
Zaptel, libpri, asterisk, all CVS from today (2004/11/08 noonish, HSV
time.) (Same thing happens with older CVS.)
After some time, which varies, audio fails to move in either direction.
I hear my own voice echoing back, at which point I know the call is no
longer usable. Call back and it works awh
They way the macro is designed you have to notify the caller in your
voicemail greeting that they need to hit pound for outcall delivery.
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> John Bittner
> Sent: Monday, November 08, 2004 1:09 PM
> To:
Interestinging
From my voicemail.conf,
my context where I
define my mailboxes in this file is
[sip]
In the sip.conf I have [EMAIL PROTECTED]
Changed that to [EMAIL PROTECTED]
and it seems to work better now.
Thanks!
Wiley
From: Paul Rodan [mailto:[EMAIL PROTECTED]
Sen
On Mon, 8 Nov 2004 15:43:10 -0500, Uma S. Pandey <[EMAIL PROTECTED]> wrote:
> In Asterisk, Can we have same extension number in different contexts?
> For example:
>
> [Context_company_1]
> exten => 200,1,,,
>
> [context_company_2]
> Exten =>200,1,..
>
> [context_company_3]
> Exten =>200,1
On Mon, 2004-11-08 at 15:43, Uma S. Pandey wrote:
> Hi
>
> For a customer, I am trying to setup 3 different companies on one
> asterisk box, and I need to assign extension 200 in three different
> companies. I was using different contexts, but was unable to get it to
> work. So, my basic question
http://www.eweek.com/article2/0,1759,1708170,00.asp
sounds all good, any body get this running with Asterisk
yet?
Cheers,
Dean
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What would one enter to get the stable or 1.x version of Asterisk and
associated modules via CVS? I've googled and wikkied but I'm using the
wrong terms or asking the wrong questions.
TIA
-Nate
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The WiSIP phone supports WEP 128 encryption. Not sure if it supports WPA
encryption, but that'd be your best bet. I'd use maximum encryption, and
separate your AP from your regular network. Just plug an AP into another
Ethernet card on your Asterisk server. The phones only need to talk to the
Aster
Paul Rodan wrote:
It actually uses 2 wires for positive and 2 wires for ground/negative? So
it's combing 2 wires (instead of 1) to deliver more power?
I believe so, although apparently there is a configuration where the
power is present on the data wires instead... I've never seen that though.
W
Cat3 - which used to be called "D Inside Wire" (DIW) *is* the wire
spec'd in the 10baseT IEEE standard. The existing wire plant is
currently to the 10baseT standard., at least as far as the wire goes.
(It was originally invisioned that 10bt and analog/digital voice would
be running in the same
It actually uses 2 wires for positive and 2 wires for ground/negative? So
it's combing 2 wires (instead of 1) to deliver more power?
Which 2 are positive and which 2 are negative/ground?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kevin P.
Fleming
S
Hi
For a customer, I am trying to setup 3 different companies on
one asterisk box, and I need to assign extension 200 in three different companies.
I was using different contexts, but was unable to get it to work. So, my basic
question is -
In Asterisk, Can we have same extension n
On 11:39 AM 11/8/2004, Paul Rodan wrote:
>Help anyone? I hate caller ID.
I would do something like this:
Set accountcode to the callerid number for each sip ua. In other words if
my callerid for a sip UA was "John F. Doe" <2025551212>, then I would set
the accountcode to 2025551212
Then I would
Hello Geraldo and Partners ,
I can offer a conference room on my * BOX on next friday to give a "start"
on this idea.
This conference will be made in "Portuguese" and will start
at 11/10/2004 At 8:00PM
If someone is interested , please contact me off list for more information.
Best Regards,
-Jeffe
On Mon, 8 Nov 2004, Mamadou Lamine KA wrote:
> Hello everybody;
>
> I would like to know the parameters on which depend jitterbuffer in
> iax.conf. Is there some kind of formula to set the correct values?
>
> Thanks in advance for any help
>
> Lamine
I'd say that the numbers in the iax.conf
Hi,
Callback is what I based my script on.
The problem I am having is when someone leaves a messages
and then hangs up, the rest of the macro does not continue
to run. If after I leave a message I hit # it works perfect.
Any ideas?
John Bittner
Simlab.net
> -Original Message-
> From
Anybody know anything about this phone:
http://www.ipeya.com/SIP_Phone_1001.htm
Other phones they sell look like Grandstream phones.
Could this be Grandstream's new phone?
Thanks,
John
--
John Gray [EMAIL PROTECTED]
AgoraNet, Inc. (302) 224-2475
102 E.
What does it show in
/var/spool/asterisk/voicemail/default/extension/INBOX/ ?
Sometimes when my users delete a message
or move them around, the sequential order in the INBOX will get thrown off. So
the phone’s light will stay on, because Asterisk can see a file(s) in
there, but when the
Have you attempted to use SIP? It's working quite well for me.
sip.conf
[maxtnt]
type=friend
host=xxx.xxx.xxx.xxx
dtmfmode=inband
callerid="MaxTNT"
context=toll-access
qualify=yes
reinvite=no
canreinvite=no
disallow=all
allow=g729
allow=ulaw
extensions.conf
(xxx.xxx.xxx.xxx would be the addres
Anyone having issues
with the message indicator lights after CVS-HEAD-07/23/04-13:55:59
??
For several of my
users, our MWI lights do not turn off. Phones are Polycom IP500 and this
just started prior to my last update.
Should I update to a
newer version? I pulled this from the CVS last w
Ok. I discovered that this flag will not work, it actually sets the caller
ID to the extension being dialed, ie:
exten => 1235551212,1,Dial(SIP/whatever,15,f)
works perfectly. The caller id will show 1235551212, however:
exten => 1212,1,Dial(SIP/whatever/15,f)
does not work. I believe it tries
Hi Denis, congratulations for the initiative.
I would be glad to help. Feel free to contact me PVT.
Regards
Geraldo
-Mensagem original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Em nome de Denis Galvão
Enviada em: sexta-feira, 5 de novembro de 2004 17:55
Para: [EMAIL PROTECTED]
My standard answer to POE questionsMostly stolen from or repeated in
a Network Computing issue about 2 months ago.
PoE factoids:
PoE uses the spare pairs *or* the data pairs (which one to use is
automatically detected) in an ethernet (10 or 10/100) cable to carry
-48V dc from the power sou
We currently have an Asterisk installation and need to add cordless /
wireless phones. Requirements are these phone need to be equals to the
"wired" devices, i.e. dedicated buttons for hold, transfer, etc. , e.g. not
an ATA connected analog phone cordless phone. Was thinking of using 802.11b
SIP
Tim Donahue wrote:
First, I will admit that I have not worked with PoE before so I'm asking
this for my own benifit as well as the OP's benifit. Doesn't PoE
require at lest 3 pairs to be availible? I know that pins 1, 2, 3, and
6 get used for ethernet communications and doesn't the power get
tran
First, I will admit that I have not worked with PoE before so I'm asking
this for my own benifit as well as the OP's benifit. Doesn't PoE
require at lest 3 pairs to be availible? I know that pins 1, 2, 3, and
6 get used for ethernet communications and doesn't the power get
transmitted over pins 4
Thanks, it is finally working.
Where can I find more info on the priorities in Asterisk?
Nicklas
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bill Hamel
Sent: den 8 november 2004 16:56
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subje
I'm having trouble with a TDM400P card configured with one fxo and one
fxs device. The system is a Mac G3 B/W running YellowDog 3.01 (2.4.22-2f
kernel).
The card is installed with a power cable, it configures itself properly
at boot time, ztcfg and zttool shows everything is fine. Asterisk 1-0-
Thanks for your efforts Steve, but it
turned out to be a problem with SJphone. X-Lite does not exhibit the same
symptoms.
Peter
From:
[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
On Behalf Of Steve Totaro
Sent: Friday, November 05, 2004
7:43 AM
To: Asterisk Users Mailing List
Have you tried chan_sccp?
http://chan-sccp.sourceforge.net
Matthew
- Original Message -
From: "James Forte" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<[EMAIL PROTECTED]>
Sent: Monday, November 08, 2004 12:20 PM
Subject: Re: [Asterisk-Users] Cisco
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> John Bittner
> Sent: Monday, November 08, 2004 10:39 AM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: [Asterisk-Users] Voicemail Macro issue.
>
> Hi,
>
> Anyone know ho
Rich Adamson wrote:
Polycom ships out two different phones , ones with H323,and one with
SIP
already loaded.
Thank you,
Steve Maroney
Correction, the polycom IP 500 ships without h.323 or SIP
software (it only has a bootrom on it), and software is only
distributed by polycom authorized VoIP partn
Hi guys,
I am so happy to hear that. I'm agree with Jefferson
There a lot of people working with * here in Brazil.
I would be happy to help you.
I have a idea, what do you think about have a chat some day in this week
with the Brazilian guys?
Geraldo Santo
Osasco-SP-Brazil
-Mensagem ori
Your question indicates that there may be a better way...
???
I want to use the voice mail and extension features of Asterisk, and
sometimes there is this NAT problem that Asterisk seems to handle very
well.
I've been using H.323 with the TNT.
Do you have an alternate solution?
On Mon, 8 Nov 2
> You could maybe look at the autocreatepeer option for sip.conf
that level of vulnerability would not seem to be a good approach
to solving some sort of sip/config problem :-)
the problem is in the sip handshake between the spa3k and *. i
have been hoping a sip geek would have a chance to look
I have two 7910's one is a 7910G+SW and one is 7910+SW
I have the 7910G+SW to work with an xml file in the /tftpboot directory.
Using chan_skinniny however I cannot get the hold tranfer etc. buttons to
work.
skinny.conf is as below:
---
501]
context=default
Michael,
Attached some of the logging.
I noticed that when I call the sip number, it surely is talking to my
ipphone. When I look at the debug info coming out of my
phone it starts to spit out information (not readable) so for sure
asterisk and the phone are talking.
I tried setting a different
Hi,
I have a * server
which does only SIP to H323 completely in IP domain, there is no digium h/w in
it. In your experience, for this type of application, is it required to have a
timing source to prevent the calls being dropped.
Cheers
SW
_
Hmmm... You're right, I must have missed that option. If this works, I do
apologize for wasting your valuable time. However, do I put this on the
outbound or inbound rule?
This rule:
; Local
exten => _9NXXNXX,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN:1})
exten => _9NXXNXX,2,Congestion
or do I
What to answer to this one?
Module loaded and no 'OH323' channel type registered?
How did you do that?
As a last attempt, enable debugging on the console (logger.conf)
and start Asterisk with -vvvcd, rerun and email the full output.
Also, send the portion of Asterisk boot messages (where it loads
t
I have a Grandstream 101 phone on my desk. I also use a Microsoft
Wireless Optical mouse.
When I'm using the phone, the mouse doesn't work very well - herky jerky
movement. If I move the phone away from the mouse and receiver, they
work fine again - otherwise I need them very close together to
Michael,
When I do show modules it shows up in the list..
And if it wasn't loaded, how come asterisks can still receive h323 calls?
Alex
apeldoorn*CLI> show modules
ModuleDescription Use Count
chan_modem.so Generic Voice Modem Drive
Hi,
Anyone know how to get voicemail to continue running the
next exten in the dialplan when a user hangs up. If a user
hits # after leaving a message instead of hanging, up it
works. I am trying to do a call back macro and when users
hangup after leaving a voicemail the rest of my macro does
not
Ok I figured it out mostly, I went
by what Flynn posted. I commented out
the particular lines in res_features.c when a call is connected no DTMF is
passed. The only problem I am
having now is I’m not sure how to set up the IVR. This is what I need to be done. Sounds simple and probably
Paul Rodan wrote:
Nobody responded so I’m sending this out again. I need help on stopping
the “Change caller ID on forward” trick that either Cisco or Asterisk
keeps doing. My upstream provider doesn’t like it.
This doesn't help?
'f' -- Forces callerid to be set as the extension of the lin
Alex van Es wrote:
Michael,
Yeah.. for sure the channel is loaded.. calling to my asterisks works fine.
I have included the oh323.conf and the original message.
Thanks a lot for you help. I would would like to get this baby working.
Alex
The log;
Nov 8 18:04:01 WARNING[294930]: channel.c:1901 ast_
I commonly use the DND feature in Asterisk by dialing *78. When I do this I
hear about a second of stutter dialtone to let me know the feature was set.
Is it possible to configure the Zap channel to continue to provide stutter
dialtone while the line is in DND? This way if someone forgets to turn
Essentially I wish to have buttons on a panel (like the Snom 220's
extension board) that show when people are on the phone or off the
phone for a receptionist.
As far as I know, you can't do this with asterisk, at least not
easily.
From what I've read, most people call this "shared lines" or
som
Nobody responded so I’m sending this
out again. I need help on stopping the “Change caller ID on forward”
trick that either Cisco or Asterisk keeps doing. My upstream provider doesn’t
like it.
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Paul Rodan
Sent: Friday
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