Hello,
I've got a Cisco ATA 186 from Vonage plugged into my Asterisk box with a
X100P card. This works great for the most part, but I'm having a
disconnect supervision problem.
I suspect the Cisco device doesn't provide any sort of analog disconnect
supervision when it gets a SIP BYE message
Is it possible to have multiple
Asterisk servers installed on a LAN?
How are the SIP phones connected to the Asterisk
servers?
Any redundancy configuration?
Anyone with these configurations
working?
Thanks.
How Peng Kaiam
When I make a call using iconnecthere, I hear no ringback tone, but the
call does get connected. Any suggestions?
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I would recommend then doing "grep fromstring
/usr/src/asterisk/apps/app_voicemail2.c"
Martin
On Fri, 19 Sep 2003, Ben Bloomberg wrote:
> I'm having tons of trouble getting the fromstring to work in
> voicemail.conf. I've tried both voicemail and voicemail2 but the emails
> still seem to be
I'm having tons of trouble getting the fromstring to work in
voicemail.conf. I've tried both voicemail and voicemail2 but the emails
still seem to be coming from asterisk pbx. Has anyone had any luck with
this?
=
Here's my voicemail.conf:
;
; Voicemail Configuration
;
[general]
;
I'm considering using asterisk to replace an existing PBX in a 40 room hotel and
would appreciate any comments, corrections or insight before I begin.
Only 8 PSTN connections are initially required but since the guests need dial-up
internet access in the rooms it has to be Frac-T1 as opposed to
I have inbound IAXtel calls working, but they come into the wrong
context.
I have a context= line in general above the register line in iax.conf
Does anyone have any ideas what might be happening?
--
Sample configs and more: http://www.fnords.org/~eric/asterisk/
BTEL Consulting
+1-850-484-4535
Hi,
I have asterisk configured for german ISDN
and SIP. SIP only for intranet connections.
In our office there is a snom 100 and a snom 200
phone.
When I'm calling a (public) telephone number
which is busy, asterisk chan_modem hangs.
Busy is never indicated to the calling SIP phone.
And afterwords
Try this link:
http://gnophone.com/directory/createAccount.php
You will find it on the setup page. The first line says: To sign up for
free IAXtel access, go here.
The word 'here' is the hyperlink to account creation form.
As for top posting, it is what I just did here.
You should also setup y
Anyone knows there exist such a software that is working with Asterisk?
Thanks
=
William Zhang
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These functions are implemented only for chan_zap (zaptel hardware) and
work for FXS/FXO ports. Exception is *8 (remote call pickup) as far as I
know.
regards
Martin
On Fri, 19 Sep 2003, Rich Adamson wrote:
> Someone recently posted the following list as functions built into *
>
> *0# sends fla
When I make a call using iconnecthere, I get no ringback tone, but after
the ringing the call does get connected. Any suggestions?
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I've been poking at the Skinny channel driver since yesterday and
noticed a few things.
There doesn't seem to be inbound audio to the 79[46]0 phones. The
status display shows a 0ms packet size. This was for calls to and from
Zap, SIP and another skinny phone. One clue was that a skinny to skinn
-Original Message-
From: Gary [mailto:[EMAIL PROTECTED]
Sent: Friday, September 19, 2003 7:49 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Radio for Music on Hold?
sometimes its more relevant to drop a caller into MOH with a special
broadcast
EG: here in cairns, we have p
I'm trying to filter calls that don't have a proper ANI. This is what
I did:
; only if they a real-looking ANI
exten=_1XX1118/_.N.,1,Newt,1118-config
; Otherwise, send them to the loser partyline
exten=_1XX1118,1,Goto(outtrunk,19096611234,1)
This properly deals with null ANIs, but for some
sometimes its more relevant to drop a caller into MOH with a special
broadcast
EG: here in cairns, we have permission during Cyclone watch etc to
rebroadcast, it would be very relevant to have users listening to the
local radio station whilst on hold during those times.
On Fri, 19 Sep 2003 06
That's my understanding!
IAX is UDP/5036
IAX2 is UDP/4569
You will probably want to use IAX2 since it can 'trunk' multiple calls
in one packet.
Let me know how it goes.
Regards,
Steve
On Fri, 2003-09-19 at 18:57, C. Johnson wrote:
> Ok so if I understand correctly:
>
> For IAX, just open up t
How do you set up IAX in Trunk mode?
Uriel
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of WipeOut .
Sent: Friday, September 19, 2003 3:49 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] IAX vs SIP
> I wonder how IAX compares to SIP bandwidth-wise? I
I forgot...
the main problem is that eu phones seems to have flash timings
~80 - ~120 ms , so with default zaptel values, a flash hook
('R' button) is received by asterisk as one pulse, since
the pulse time is set up to 150ms ...
Matteo.
Il sab, 2003-09-20 alle 01:15, Brancaleoni Matteo ha scritt
Hi.
zaptel.h , line 789
#define ZT_DEFAULT_RXFLASHTIME 1250
For italy I had to lower it to 200,
also be sure to lower the pulse timer
(unless you're using a pulse phone with asterisk)
line 792
#define ZT_MAXPULSETIME (150 * 8)
I moved it to (20 * 8)
be sure not to set it under ZT_MINPULSETIME, t
Ok so if I understand correctly:
For IAX, just open up the IAX ports on the firewall (the exact
numbers escape me right at the moment), and let it fly?
-cj
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Stephen Varga
> Sent: Friday, September
> ClientServer
>
> XTEN <--> */Firewall(NAT) <---IAX---> Firewall(NAT)/*
>
If you are going to use IAX, I don't think you have to put * on the
firewall boxes, only if you wish to use SIP.
Steve
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Asterisk-Users mail
Someone recently posted the following list as functions built into *
*0# sends flash
*8# remote call pickup (pickup phone in your group)
*67# disable caller id
*70# no call waiting
*78# do not disturb on
*79# do not disturb off
*72# enable call forwarding
*73# disable call forwarding
*82# enable
Yep.. You are right...
Hmm
It is sending the SIP client 10.xx.xx.15 which is the local IP addr
somewhere
else.. darnit.
From: "asterisk" ;tag=as3aff06a5
To:
Contact:
Ok, so now, I guess my network layout has to be this:
Client Server
XTEN <--> */Firewall(NAT) <---I
-=> On Fri, 19 Sep 2003 15:39:44 -0500, PJ Welsh <[EMAIL PROTECTED]> said:
> I am about to try our TDM400P E model from the developer kit (not
> the "Lite") we just got and noticed a large number of reported
> problems. I had the CVS from Sep 12 (or so the CVS/Entries file has
> in it). My drivers
I'm trying to filter calls that don't have a proper ANI. This is what
I did:
; only if they a real-looking ANI
exten=_1XX1118/_.N.,1,Newt,1118-config
; Otherwise, send them to the loser partyline
exten=_1XX1118,1,Goto(outtrunk,19096611234,1)
This properly deals with null ANIs, but for so
Don't know yet if it helps, but if you read the link at:
http://www.voip-info.org/tiki-index.php?page=NAT+and+VOIP
it will point you to:
http://www.sipcenter.com/files/SIPNATtraversal.pdf
However has the voip-info.org site; your stuff ROCKS!!
On Fri, Sep 19, 2003 at 03:11:31PM -0500, C. Johnso
I am about to try our TDM400P E model from the developer kit (not the "Lite") we just
got and noticed a large number of reported problems. I had the CVS from Sep 12 (or so
the CVS/Entries file has in it). My drivers seem to modprobe fine. My card show up as
"Found a Wildcard FXS: Wildcard S400P
I am new to * and I have been attempting to solve this same issue, but
have come to the conclusion that they only way to make it work is for *
to have a real reachable IP address or place another * box at the second
site and use IAX trunking. This second * box, unfortunately is
unsuitable for my sc
> I'm having a problem where the recall button doesn't work
For the benefit of our non-UK readers, recall = flashhook, but usually only
100ms timed line break, not the 500ms which seems to be the norm in North
America.
Robb - am I right in saying your using a UK phone? And it's definitely using
ti
Hi capi users,
chan_capi 0.2.5b brings some new options to simplify your capi.conf.
msn= is now a comma separated list of allowed outgoing MSNs
controller= takes a comma separated list too (no need to copy your
capi interfaces for multiple controllers anymore)
incomingmsn=* catches every incomi
Hello Folks-
Pretty new to the list here, got a lot of reading to do.. Does anyone
know where I can find a decent HOWTO or set of instructions for
running
Asterisk and SIP clients thru firewall/NAT systems?
I have a Asterisk box sitting behind a linux firewall at a remote
location
and have the 50
Hi
I'm having a problem where the recall button doesn't work
If i press recall before I dial numbers it disconnects me which is what
I would expect, but during a conversation if I want to transfer the TDM
400 just ignores the recall
Any advice would be gratefully received
Thanks
Robb
_
I'm doing the following to integrate * and a Partner ACS using an 8x16
Zhone channelbank.
Channel 1-4 => FXS (extensions) on Partner
Chanenl 5-8 => POTS/PSTN
Channel 9-16 => FXO (CO lines) on Partner
This setup is working pretty well, except for a few issues with call
supervision on the Zhone.
On Fri, 19 Sep 2003, WipeOut . wrote:
> > I wonder how IAX compares to SIP bandwidth-wise? I've tried both over
> > overseas IP connection, and somehow SIP seemed to work better.
> >
> > Peter
> >
>
> Then try making two or three or more calls at the same time.. :)
>
> If you setup IAX in tr
Does this thread help?
http://lists.digium.com/pipermail/asterisk-users/2003-June/014788.html
On Fri, Sep 19, 2003 at 01:18:53PM -0500, Peter Zeltins wrote:
> I wonder how IAX compares to SIP bandwidth-wise? I've tried both over
> overseas IP connection, and somehow SIP seemed to work better.
>
> I wonder how IAX compares to SIP bandwidth-wise? I've tried both over
> overseas IP connection, and somehow SIP seemed to work better.
>
> Peter
>
Then try making two or three or more calls at the same time.. :)
If you setup IAX in trunk mode it uses the same connection for multiple voice str
> I looked at the gnophone web but it does not tell
> me how to register or where!
>From the main page, click on Setup (at the top), then there's a link to
create a new account. Or just click here:
http://gnophone.com/directory/createAccount.php
> Also can someone explain what top posting is? I do
Ok I know that I am new user and would like some information on how to register to use
IAXTEL. I looked at the gnophone web but it does not tell me how to register or
where! Yes I am very new to Asterisk and Linux so please help. I did a google on this
and kept sending back to the main site IA
I've had problems with Dialogic apps using GlobalCall with similar
symptoms, I had to type "export LD_PRELOAD=/usr/dialogic/lib/libgc.so"
before running them. Maybe Mark's answer solves that problem also...
Tim
> Need to have chan_dialogic.so => yes in the [globals]
>
> Mark
>
> On Thu, 18 Sep
> I have that line in my iax.conf
Are you using the password they gave you when you signed up, or the new
password that you were forced to pick when you logged in for the first time?
I think the screen says something about it not changing your IAXTEL
password, just the one you log in to the web sit
> > > I have a machine in my office, it is labelled Cable and Wireless, and on
> > > the back it says, SMarT-1
> > > I have searched the web, and no joy. It connects to a PC via a serial
> > > cable, has anyone heard of such a device?
> >
> > Sounds like a CSU/DSU that C&W installed for some servi
> I'm trying to interface * with a PBX, but seems that his
> ring cadence is somewhat different, and my T100 doesn't
> show any call coming in.
Yeah, I had a similar problem - I was trying to connect an X100P to a small
3x8 analog PBX for testing and it wouldn't grab the call. Thinking about it
now
I wonder how IAX compares to SIP bandwidth-wise? I've tried both over
overseas IP connection, and somehow SIP seemed to work better.
Peter
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One more : http://www.zipworld.com.au/~erikd/XMMS/
This one uses libsndfile : http://www.zip.com.au/~erikd/libsndfile/
which can play even more formats including gsm6.10, G721 & G723 (quite
impressive)
On Fri, 2003-09-19 at 18:41, marrandy wrote:
> Hello.
>
> I can't find a gsm plugin for XMMS.
Need to have chan_dialogic.so => yes in the [globals]
Mark
On Thu, 18 Sep 2003, pedro bulach gapski wrote:
> I am one of those trying to use old dialogic hardware with *. I have the
> following error when loading the driver:
> [chan_dialogic.so] => (Dialogic Global Call API Support)
> dlopen of
You may try this one :
http://www.68k.org/~michael/xmms/
It uses the audiofile library that plays many formats.
On Fri, 2003-09-19 at 18:41, marrandy wrote:
> Hello.
>
> I can't find a gsm plugin for XMMS.
>
> How do Unix, Linux, BSD users listen to gsm samples ?
>
> Regards...Martin
--
:: M
> Hello.
>
> I can't find a gsm plugin for XMMS.
>
> How do Unix, Linux, BSD users listen to gsm samples ?
>
I just use playback in Asterisk..
--
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Now with e-mail forwarding for only US$5.95/yr
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___
>
> At the risk of sounding stupid. what's CSU/DSU ? *i'm googling it
> right now, but it's nice to have convo on the list!*
A CSU/DSU is Channel Service Unit (CSU) this terminates T1 connections
from the phone company. This information is then passed to the Data
Service Unit which turns th
Have you tried starting asterisk with -c? It should give you some
detail as to what is happening with the call.
Scott M. Stingel
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Thomas Haeger
> Sent: Friday, September 19, 2003 3:40 PM
> T
Hi,
I was trying to test the conferencing application, here is my setting in the extensions.conf
exten => 5,1,MeetMe,44|p
and my meetme.conf is
conf => 44
but when i press the # , it doesn't exits my line from the conference, any suggestions
Regards
Azher
Do you Yahoo!?
Yahoo! SiteBui
hi.
from a shell, just type : play filename.gsm
matteo.
Il ven, 2003-09-19 alle 18:41, marrandy ha scritto:
> Hello.
>
> I can't find a gsm plugin for XMMS.
>
> How do Unix, Linux, BSD users listen to gsm samples ?
>
> Regards...Martin
--
Brancaleoni Matteo <[EMAIL PROTECTED]>
Espia - Emmegi
>So, do you have the P/S 4-way connector plugged into the TDM400P ?
>Regards...Martin
Yes, and I've tested the voltage to the card. Both the 12V and 5V supplies
are OK to the card.
-Sean
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Hello.
I can't find a gsm plugin for XMMS.
How do Unix, Linux, BSD users listen to gsm samples ?
Regards...Martin
--
While you don't greatly need the outside world, it's still very
reassuring to know that it's still there.
___
Asterisk-Users mailing
On Fri, 2003-09-19 at 10:58, Dante Alzamora wrote:
> Howdy,
>
> I need some pointers (ideas or help) to build a solution to retrieve
> recordings out of an IVR.
>
> The program needs to do some predictive dialing functions I only need it to:
>
> 1) Be able on it's own to make a call (to the same
On Fri, 2003-09-19 at 10:53, Scott Stingel wrote:
> Mark-
>
> Yes, you can create a shell script that dumps a text file into
> /var/spool/asterisk/outgoing.
>
> Use the prototype found in /usr/src/asterisk/sample.call
>
> Name the file "N.call" or something similar, where N is the channel number
Doesn't matter it should still work. Here is a hint.. dont use
passwords/secrets it will then work!
bkw
On Fri, 19 Sep 2003, Xisco wrote:
> That's true if always there to connect two asterisk servers, but I'm doing
> some proves in order to connect one asterisk server with another SIP server.
>
Mark,
I added one to the bug report. Hope that helps.
bkw
On Fri, 19 Sep 2003, Mark Spencer wrote:
> I"ll need a backtrace.
>
> Mark
>
> On Fri, 19 Sep 2003, Dave Cotton wrote:
>
> > Using CVS update from 11:00 CET today * crashes at this point.
> >
> > == Parsing
> > '/var/spool/aster
Come on you guys! Steven is right. It took me all of two minutes to
find the manual for this little bvox using google
Find it on this page http://www.dialerbuddy.com/mitel.htm
How to use it from Linux? It's a serial device send data over the
serial port. RTFM.
--- Angel Gabriel <[EMAIL PRO
> > Hi all,
> >
> > Can Asterisk **initiate** a call?. If yes, what is the command?
> >
> > I would like that Asterisk automatically calls to me (or to
> somebody) and
> > reproduces a mp3 locution, a menu, etc., is it possible?
>
> Try using Dial...
>
> From the console type 'show applicati
I have an Aastra 390 ADSI phone. It's not locked.
I can call ADSIProg without a problem and it programs my phone. Calling
Voicemail2 also programs my phone.
However, in order for the VMail option to appear on the screen I have to
go into the Services menu, pick Asterisk PBX and pick Select.
Th
Howdy,
I need some pointers (ideas or help) to build a solution to retrieve
recordings out of an IVR.
The program needs to do some predictive dialing functions I only need it to:
1) Be able on it's own to make a call (to the same number inside this
script).
2) Detect that the call has been answ
On Fri, 2003-09-19 at 17:14, Rich Adamson wrote:
> > I have a machine in my office, it is labelled Cable and Wireless, and on
> > the back it says, SMarT-1
> > I have searched the web, and no joy. It connects to a PC via a serial
> > cable, has anyone heard of such a device?
>
> Sounds like a CSU/
Mark-
Yes, you can create a shell script that dumps a text file into
/var/spool/asterisk/outgoing.
Use the prototype found in /usr/src/asterisk/sample.call
Name the file "N.call" or something similar, where N is the channel number.
Create an outgoing context in your extensions.conf file to do wh
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] Behalf Of Cerrajetto
> Sent: Friday, September 19, 2003 11:35 AM
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] Can Asterisk automatically initiate a call?
>
>
> Hi all,
>
> Can Asterisk **initiate** a call
On Fri, 2003-09-19 at 10:34, Cerrajetto wrote:
> Hi all,
>
> Can Asterisk **initiate** a call?. If yes, what is the command?
>
> I would like that Asterisk automatically calls to me (or to somebody) and
> reproduces a mp3 locution, a menu, etc., is it possible?
Look at sample.call in the source
On Fri, 2003-09-19 at 09:36, Angel Gabriel wrote:
> I have a machine in my office, it is labelled Cable and Wireless, and on
> the back it says, SMarT-1
> I have searched the web, and no joy. It connects to a PC via a serial
> cable, has anyone heard of such a device?
This isn't a flame, but maybe
Hi Folks,
I'm trying to interface * with a PBX, but seems that his ring cadence is
somewhat different, and my T100 doesn't show any call coming in.
I've tried to change zaptel to new values but still couldn't make it
work.
Is there any other place where I should be changing some parameter? Is
th
Hi,
The following equipment is forsale on ebay:
Wildcard T100P (two weeks old):
http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&category=51279&item=3048079393
Adtran TSU 600 with 12 FXO ports:
http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&category=44993&item=3048077400
Cisco 7940 loaded with v5.3
Hi all,
Can Asterisk **initiate** a call?. If yes, what is the command?
I would like that Asterisk automatically calls to me (or to somebody) and
reproduces a mp3 locution, a menu, etc., is it possible?
Thank you,
Mark
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Asterisk-Users mailing list
On Friday 19 September 2003 11:04 am, Sean Rodger wrote:
> There is also an additional problem now of the driver occasionally flooding
> my screen with kernel error messages. These are different error messages
> than the original "Power alarm on module N, resetting!", (sorry I don't have
> the new
On Fri, 2003-09-19 at 17:14, Rich Adamson wrote:
> > I have a machine in my office, it is labelled Cable and Wireless, and on
> > the back it says, SMarT-1
> > I have searched the web, and no joy. It connects to a PC via a serial
> > cable, has anyone heard of such a device?
>
> Sounds like a CSU/
On Fri, 2003-09-19 at 10:04, Sean Rodger wrote:
> Can anyone tell me if they have had any problems using the Digium X100P
> cards and the
> Cisco ATA186 together with asterisk??
Yes. The only codec that is compatable with Asterisk without additional
non-free codecs is the ULAW or ALAW codec. Se
Hi all,
Thank you for your help, finally we have found that it was a codec problem,
now both systems are forced to use g711 ulaw and outbound calls are working
fine.
Best regards,
Mark.
-- Original Message ---
From: "Cerrajetto" <[EMAIL PROTECTED]>
To: [EMAIL PROTECTED]
Sent:
> I have a machine in my office, it is labelled Cable and Wireless, and on
> the back it says, SMarT-1
> I have searched the web, and no joy. It connects to a PC via a serial
> cable, has anyone heard of such a device?
Sounds like a CSU/DSU that C&W installed for some service.
Here is some more information about my problem:
With 2 phones plugged into the 4 port FXS card, here is a situation I have
witnessed:
I have a clean dialtone one phone. The instant the other phone goes from
on-hook to off-hook, the clean dialtone on the first line turns into a loud
crackling soun
I have a machine in my office, it is labelled Cable and Wireless, and on
the back it says, SMarT-1
I have searched the web, and no joy. It connects to a PC via a serial
cable, has anyone heard of such a device?
--
*
Not everyone is touched by an Angel
Those that are, never forget the
Hi!
I was trying to use gnophone with asterisk, but I can't make a call (It
just get the a answer of "REJET"), but I can register an everything.
Anyway, I decided to move to the cvs version of gnophone, so I checked
out EVERYTHING from cvs.digium.com (yes, a cvs -z7 co .). I installed
libiax
Hi all,
i tried to make a call from public pstn in our */E100P.
Config is following:
exten => _X.,1,Playback(testgsm)
But what i hear is one dtmf tone and then nothing...
Any ideas ?
Regards,
Thomas.
***
beroNet technologies GmbH
Dipl.- Ing. Th
I have a machine in my office, it is labelled Cable and Wireless, and on
the back it says, SMarT-1
I have searched the web, and no joy. It connects to a PC via a serial
cable, has anyone heard of such a device?
--
*
Not everyone is touched by an Angel
Those that are, never forget the
- Original Message -
From: iPfone Telefonia
IP
To: [EMAIL PROTECTED]
Sent: Friday, September 19, 2003 11:27 AM
Subject: hangup problem Brazil
Hi
all!I´m setting up an asterisk box here in brazil, asterisk don´t hangup
afterthe caller disconects...it goes to voice mail etc.. Som
I have that line in my iax.conf
--- Rich Adamson <[EMAIL PROTECTED]> wrote:
>
> > Has anybody had a problem registering their IAXtel
> > account?
>
> My account is working fine using the following in
> iax.conf:
> register => username:[EMAIL PROTECTED]
> towards the bottom of the [general] s
> Has anybody had a problem registering their IAXtel
> account?
My account is working fine using the following in iax.conf:
register => username:[EMAIL PROTECTED]
towards the bottom of the [general] section.
(I didn't test indial as of this morning to actually validate, but it
was working pri
Has anybody had a problem registering their IAXtel
account?
I just signed up for an account and followed the
documentation on iaxtel.org and my registration is
always rejected.
When I type "iax show registry", I get the following
output:
Host UsernamePerceived
Ref
Yep Dave same here. It segfaults just as the digit playback starts. This
is true even without tz= options set.
Holds true with 'make clean' 'make update' 'make' 'make install'.
For those that need voicemail, beware. :)
Regards,
--- Gavin
> -Original Message-
> From: Dave Cotton [mailto
I"ll need a backtrace.
Mark
On Fri, 19 Sep 2003, Dave Cotton wrote:
> Using CVS update from 11:00 CET today * crashes at this point.
>
> == Parsing
> '/var/spool/asterisk/voicemail/default/2201/INBOX/msg.txt': ==
> Parsing '/var/spool/asterisk/voicemail/default/2201/INBOX/msg.txt':
>
Using CVS update from 11:00 CET today * crashes at this point.
== Parsing
'/var/spool/asterisk/voicemail/default/2201/INBOX/msg.txt': ==
Parsing '/var/spool/asterisk/voicemail/default/2201/INBOX/msg.txt':
Found
Sheriff*CLI>
Disconnected from Asterisk server
--
Dave Cotton <[EMAIL PROT
Hi all,
can somebody explain me why i can't hear a ringing tone (alerting) if i'am
going to connect to my destination end point?
Is it basically so that i have to configure like:
exten => xxx,1,Dial,ChanTec/number|timout|r
Is it really nessesary to use the "r" option everytime if i want
Look at all the time you are wasting flaming people. just ignore these
questions and get off the high horse. Do you maintain this list? If not
then you have no say whatsoever.
- Original Message -
From: "Steve Creel" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Thursday, Septembe
On Thu, Sep 18, 2003 at 01:21:54PM -0700, Paul Crick wrote:
> Come on people! Fork out $50 for a discman and another few bucks for some
> royalty free library music and have that on hold instead.. You're in
> control, you know what your callers are listening to, and you're also legal
Why go to all
"Olle E. Johansson" <[EMAIL PROTECTED]> writes:
>>>don't now and simply add "What's a pyroflax?" on it. Someone will
>>>notice and explain what a pyroflax is...
>> A what ? :-)
> Google ;-)
No way, even google is moot on that word. I guess you'll have to
explain :-)
--
Rémi
___
> Does asterisk know when each ring comes in or just the first ring, ie
> so the cadence can be worked out? say over two rings?
>
> Robb
> Martin Pycko wrote:
>
> >The X100P together with asterisk does not support the distinctive ringing
> >detection on the line. Asterisk however can generate t
On Thu, 18 Sep 2003 13:21:54 -0700, Paul Crick wrote
> > Tell your client that some callers put on hold may
> > know about the above and "radio on hold" would make
> > the company look at best ignorent.
> I read something somewhere.. can't remember where.. some PBX
> buyer's guide maybe? ANYWAY..
You would have to buy a g723.1 license which would bust every users budget :)
g723.1 is a prpriatory codec and there is no legal implementation for
asterisk.
On Friday 19 September 2003 1:11 pm, Thomas Haeger wrote:
> Hi all,
>
> i don't know how often someone ask for this, but i ask agian:
>
>
Hi all,
i don't know how often someone ask for this, but i ask agian:
Is it possible to use G723.1 with * or not ?
I tried to use G723.1 from * over OH323 to a gatekeeper from my provider.
The situation is following:
Zap/analog ---> IAX -INTERNET-IAX--->OH323>GATEKE
Your statement:
''Also a reminder to those who know far more than I, You too started
someplace and someone answered your questions and you learned''.
Very well said.
Almost always, bad and irritable manners are symptoms of deep trauma in
one's life.
A little tolerance goes a long way.
Cheers,
Ab
That's true if always there to connect two asterisk servers, but I'm doing
some proves in order to connect one asterisk server with another SIP server.
That's the matter.
- Original Message -
From: "Jamie Carl" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Friday, September 19, 2003 1
Why?
Use IAX2, it is s much better...
J
On Fri, 19 Sep 2003 11:54:23 +0200
"Xisco" <[EMAIL PROTECTED]> wrote:
Hi everybody,
I'm trying to SIP register between two asterisk, each one
have a Public IP. Asterisk told me that Unathorizae
In * one sip.conf
register =>usuario1:pass1@
In
Hi everybody,
I'm trying to SIP register between two asterisk,
each one have a Public IP. Asterisk told me that Unathorizae
In * one
sip.conf
register
=>usuario1:pass1@
In * two
sip.conf
[usuario1]
type=friendusername=usuario1
secret=pass1host=dtmfmode=i
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On Thursday 18 September 2003 19:04, Hielke Christian Braun wrote:
> try to change [siptestphone] to [atrg613test] in sip.conf. Maybe
> that helps.
It didn't. And now something else is weird. Asterisk fails sending audio to my
SIP phone. Found this
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