Does anyone know if this needs any special modification to work
outside the US? I have setup my country's correct tone info and
tested thru the indication.conf file.
Question would be, where does my zaptel device get the tones expected
for the busydetect procedure? How can I modify them? Is this
Hi,
We have a setup of the Nufone Implementation of H323 on our Asterisk Setup and
it appears to work fine apart from one slight technical glitch.
When a customer makes a call, it keeps trying to forward the client into the
default context inspite of a context=blah in the particular config.
Any i
Jeb Campbell <[EMAIL PROTECTED]> writes:
> By far the best poe (price/performance) I have seen for Cisco poe (or
> standard poe) is the Netgear
> FSM7326P. http://www.cdw.com/shop/products/default.aspx?EDC=568864
>
> It is a managed layer3 poe switch (24 port) with 2 gigabit ports also.
>
> Work
On 16/11/2004 01:58 Eric Wieling said the following:
Dinesh Nair wrote:
i do not believe that digium sells the g729 codecs for freebsd.
however, i too am a freebsd user, and i guess what is needed is more
people telling digium that we need the g729 codec on freebsd.
They do. It's considered "un
I just came across this phone MTA 3308 By Innomedia its documentation says it
supports MGCP 1.0 and SIP 2.0 anyone have any experience with this phone. I
specifically wanted to find out if it works with asterisk before i purchase
it. I checked the wiki and the lists dosent seem like anyone has tri
On Mon, Nov 15, 2004 at 10:02:43AM -0800, Bob Knight wrote:
> Håkan Källberg wrote:
> >I want to trunk two Asterisk systems with each other. System A,
> >behind a NAT-Firewall and System B with a real IP address.
> >
> >aix.conf on B:
> >
> >[mytrunk]
> >host=dynamic
> >username=mytrunk
> >auth=md5
Sorry about that ...
Anyone know how to use this feature. 'leavewhenempty = yes'
I got it to leave the queue if no one is logged in, but I
expected it to go to priority n+101
I need it to work like this.
exten => s,1,Answer
exten => s,2,SetMusicOnHold(random)
exten => s,3,DigitTimeout,5
exten =
Hello, I am having trouble with call files. I want my call files to attempt
only 1 time, and never retry. I am trying to bridge two calls together, one
call to my office [9726172877] and the other call to my cell [2022463521] My
call file looks like this:
Channel: IAX2/outgoing/19726172877
Se
Kind of looks like iaxtel is not functional again. Tried calling the
number you provided and tried calling my 700 number. Both fail with
no indication of what might have happened.
> As a test I'm trying to call the Asterisk shipping dept at (700) 428-6004
> and have had no
Yup, has the IDLE in there from the beginning.
Does the adsi.conf have anything to do with it? (Probably no, but I'm
grasping.)
On Mon, 2004-11-15 at 21:05, [EMAIL PROTECTED] wrote:
> Make sure your ADSI script has an 'IFEVENT IDLE THEN' block...
>
> IE:
>
> IFEVENT IDLE THEN
> CLEAR
>
folder. No supplier gets a purchase if their people are not properly
trained in e-mail communication. My employer spends quite a bit as
You are kidding, right? "Properly trained"? By whose standards? What
international commerce committee on email standards published the
training regimen of whic
Make sure your ADSI script has an 'IFEVENT IDLE THEN' block...
IE:
IFEVENT IDLE THEN
CLEAR
SHOWDISPLAY "titles" AT 1
SHOWKEYS "sd_1"
ENDIF
- Original Message -
From: "Rob Emanuele" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<[EMAIL PROTECTED
All,
Ok ignore that please. I was stupid and left my firewall blocking the SIP
port. Sorry for the trouble.
Leon
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Leon Oosterwijk
> Sent: Monday, November 15, 2004 10:41 PM
> To: 'Asterisk User
On Sun, 14 Nov 2004 23:22:55 -0500, Gregory Junker
<[EMAIL PROTECTED]> wrote:
> If I want to address individual points in turn I happily trim and
> inline. To say that "top posting is unprofessional" is simply a
> meaningless blanket statement; in your opinion it may be, but I doubt
> it's your mai
All,
I had asterisk and stanaphone working together at some point but now it is
not working. When I do a 'sip debug' on the asterisk server it keeps
restransmitting the packet. When I do a network trace on the sip packets
from asterisk and compare them to the sip packets generated by the
stanapho
I have a client who currently has a Toshiba PBX. We are trying to replace it with an Asterisk system. One of the features that they have on their current PBX is the ability to select a POTS line by pressing a button on their phones. They have 10 POTS lines and they can select any line by jus
Anyone know how to use this feature. 'leavewhenempty = yes'
I got it to leave the queue if no one is logged in, but I
expected it to go to priority n+101
I need it to work like this.
exten => s,1,Answer
exten => s,2,SetMusicOnHold(random)
exten => s,3,DigitTimeout,5
exten => s,4,ResponseTimeout
Steve Prior wrote:
As a test I'm trying to call the Asterisk shipping dept at (700) 428-6004
and have had no success. Is IAXTEL working? I have had no problems
with the Inter Asterisk connection part of the Asterisk demo.
It's not working for me...
--
Cheers,
Matt Riddell
As a test I'm trying to call the Asterisk shipping dept at (700) 428-6004
and have had no success. Is IAXTEL working? I have had no problems
with the Inter Asterisk connection part of the Asterisk demo.
Is there anything wrong with what I've got below?
Here is the log (password substituted of cou
Michael Vogel wrote:
[EMAIL PROTECTED] schrieb:
I suggest you take a look at http://lartc.org/
I already found a mailinglist called this name when "googleing" for
help. But I haven't found that page until now.
There you can find all the info on creating a box for shaping traffic.
Great.
Curren
OK, so I've programmed the default adsi script into slots 1 and 2.
Comedian took up slot 3.
Comedian now works properly and doesn't ask me every time to re-download
the scripts.
Going off-hook shows me the correct menu. That works right.
I still cannot get the default display to work. When t
Hello!
Can i only use one gatekeeper in OH323? Is there any documentation about
how to use gatekeeper-ids?
Thanks,
Roland
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update
The following is taken from the Sayson/Aastra 390/480 ADSI Programming Guide
Slot 1
FDN: 0.0.0.F
SECURITY: 9B.DB.F7.AC
Slot 2
FDN: 85.EF.D9.DA
SECURITY: 78.92.1D.49
Slot 3
FDN: 7B.C6.45.0C
SECURITY: 9B.60.94.30
Slot 4
FDN: FE.2E.A5.D1
SECURITY: 79.A9.0C.F0
Explanation of the Slots in Services
Slot
Not the solution.
I said I want that the calling party hears the sound always from the
beginning at start. M option enable Music on hold during dial. With Music on
hold, the listener "jumps" into the sound at random place, not the
beginning. All Callers hears the same second at the same moment. Th
I'm not sure about multiple users on the same machine.
It is just to use like a COM Object.
1. Start Skype
2. Intiate the call (with signal conversion and with error codes), once
successful
3. Capture the sound from the audio hardware buffer.
4. Convert it to the any respective codec which I presu
Sorry about taking so long to get back
The problem still exisits
I have tested both cards and they are OK
Still cant get both cards to work
r2d2*CLI> zap show channel 1
Channel: 1
File Descriptor: 21
Span: 1
Extension:
Dialing: no
Context: pstnin
Caller ID:
Caller ID name:
Destroy: 0
InAlarm: 1
From: dean collins [mailto:[EMAIL PROTECTED]
Sent: Monday, November 15, 2004 5:09 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Skype API release
http://www.eweek.com/article2/0,1759,1722578,00.asp
does this mean that Skype can now be developed to interfa
It was mentioned in the included email that an updated patch would be
released during the weekend, will this patch be re-issued by Broadvoice
or is the revised patch elsewhere?
-Original Message-
From: Rich Adamson [mailto:[EMAIL PROTECTED]
Sent: Friday, November 12, 2004 11:43 AM
To: As
Rob Emanuele wrote:
So load asterisk into Slot 1 & 2 and put Commedian in slot 3? Sounds easy
enough. Does that amke voicemail notification work too?
Where did you guys get the codes for slot 3? I have the FDN/SEC codes
for Slot 4, but have not been able to find the codes for 1,2,3.
I've talk
Roland Zagler wrote:
Could there be an issue unsing a system with Kernel 2.4.21? I had a look
at bugs.digium.com but didn't find any useful entries.
Sorry, can't reply to the rest of your mail, but I'm using 2.4.21 with
FC1 here, no probs.
--
Cheers,
Matt Riddell
_
ï
Hoi about having the calls forwarded by your phone
company?
Usually you can dial *21*number# or something and
your calls go to a remote party.
Same goes for delayed forwarding
*61*
Rene Kluwen
Chimit
- Original Message -
From:
Jim Dossey
To: Asterisk Users Maili
Ok, but is it not "alert-info" in the rfc? Dash or underscore?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Olle E.
Johansson
Brian West wrote:
> Ok to cut confusion here
>
> Its:
> Variable: _ALERT_INFO
In CVS head.
In stable, it's still ALERT_INF
Doesn't look like it -- the API is fairly limited and only available for
Windows devices at this time... and to top it all off, it's based on
Windows messages, which are inherently unreliable and slow. Skimming
through 28 pages of documentation, I have not been able to find a way to
redirect audio
Hello,
I use Asterisk 1.0.2 on a RedHat Enterprise Server 3.0 (Kernel 2.4.21)
and i experienced that the memory consumption of the asterisk-process
started by the init.d-script raises continously. Now, after 3 hours of
operation (on our testing-system we have 30 concurrent connections to
another a
> How remote are the remote offices? Miles? States? Countries? Best of my
> knowledge, the days of exchanges based on proximity to a particular CO
> are over, and those numbers (assuming they are in the same area code)
> often can be routed anywhere. You could also look into having a company
> like
Leandro Morgado wrote:
identify this as indication that the call has hung up. I'm pretty sure
the FXO module is detecting hangups through battery drop and not by
listening for busy tones, but I could be wrong here.
You can set it to detect hangups via tones in zapata.conf with the
busydetect=yes
- Original Message -
I would like to change
this to a more friendly greeting, like "Thank you for calling Comedian
Mail" and instead of it saying Mailbox have it say enter your mailbox
please.
/path/to/asterisk/sounds.txt gives you the sound file name and the text
of the file.
if you wat
Matt Riddell wrote:
Leandro Morgado wrote:
[EXTENSIVELY SNIPPED]
zaptel.conf:
loadzone=fr
defaultzone=fr
zapata.conf:
---
busydetect=yes
busycount=7
[EXTENSIVELY SNIPPED]
1. Does the Nokia 32 GSM provide you with a hangup tone? (I.E. beep
beep beep once it has hung up)
Yes it
http://bugs.digium.com/bug_view_page.php?bug_id=0002639
Thats what you want.
bkw
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Régis MARTIN
> Sent: Monday, November 15, 2004 4:36 PM
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Us
Also interesting comment about Skype possibly being interfaced directly
into a gaming solution for online game chats, does anyone know if
Asterisk has been licensed to offer something similar? Sounds like an
area that could be worth investing in.
Sort of overkill considering the popularity of pr
Peter Osborne wrote:
Hi all,
I am using the Asterisk Manager API to originate calls and it is working well,
when a call is placed the local phone rings, once you pick it up you can here
the call ringing the other end. Now, I am using Polycom IP 300 and I have
them setup to auto-answer if I set t
http://www.eweek.com/article2/0,1759,1722578,00.asp
does this mean that Skype can now be developed to interface
into Asterisk?
Also interesting comment about Skype possibly being
interfaced directly into a gaming solution for online game chats, does anyone
know if Asterisk has been
How remote are the remote offices? Miles? States? Countries? Best of my
knowledge, the days of exchanges based on proximity to a particular CO
are over, and those numbers (assuming they are in the same area code)
often can be routed anywhere. You could also look into having a company
like Voice
router. Will he be able to start downloading/uploading on that bandwidth
even though its hooked directly to the Asterisk server? If so, how can I
prevent the bandwidth usage but still allow VoIP calls?
If you do not route IP traffic over the T1 then there is no way anyone
can upload or download.
I have a client who asked me about a situation they have. They have a main office and 3 remote offices. We are installing an Asterisk server at the main office with SIP phones in the remotes. Each remote office only has 1 person. The remote offices currently have a POTS line that has a publ
On Mon, 2004-11-15 at 15:23 -0700, Kyle Hagan wrote:
Im trying to use the auto-dial out in asterisk and having problems
I get the following errors:
Nov 15 15:14:37 WARNING[1106140080]: pbx_spool.c:182 apply_outgoing:
Unknown keyword ' Channel' at line 5 of /var/spool/asterisk/outgo
Henry Devito wrote:
Hi, I have looked through the sounds files, but I think I am
overlooking the file that speaks Comidian Mail. I would like to change
this to a more friendly greeting, like “Thank you for calling Comedian
Mail” and instead of it saying Mailbox have it say enter your mailbox
Background plays a sound while waiting for an extension. You want to
Playback your short announcement then start your dial command with the m
option.
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-us
Here is our scenario:
Company A is looking for a better VoIP/Internet solution. We approach and
say, "We will sell you 1 T1 for Voice only for XXX dollars per month." They
agree. We install a router and the T1 at their location and plug the T1
directly into Asterisk. Lets say they've got a hot-s
Régis MARTIN wrote:
Hi,
I know the question was already asked but I never found an answer to
this problem. So, I try again… (things changes J)
Is there a way to play a specific message or sound from the start during
the dial command.
I want to do exactly the same thing that the “m” option of
Hi, I have looked
through the sounds files, but I think I am overlooking the file that speaks
Comidian Mail. I would like to change
this to a more friendly greeting, like “Thank you for calling Comedian
Mail” and instead of it saying
Mailbox have it say enter your mailbox please.
Kyle Hagan wrote:
Im trying to use the auto-dial out in asterisk and having problems
The error message is saying that you have a space before each command in
that file. Remove the spaces and you should be fine.
--
Cheers,
Matt Riddell
___
http://ww
I had the same problem last week. The error's are because there are extra
spaces in the 1.call file.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kyle Hagan
Sent: Monday, November 15, 2004 4:24 PM
To: Asterisk Users Mailing List - Non-Commercial Discus
As a little testbed asterisk project, to get my company comfortable with
the idea of linux telephony, I decided to ask undertake a rather small
project, replace our two boomboxes, which current supply our on-hold music
via CD and a 1/8" jack to RJ12 plug cord, and instead use Asterisk to do
it via
Hi,
I know the question was already asked but I never
found an answer to this problem. So, I try again… (things changes J)
Is there a way to play a specific message or sound
from the start during the dial command.
I want to do exactly the same thing that the “m”
option of Dial comman
On Mon, 2004-11-15 at 22:09, Peer Oliver Schmidt wrote:
> Hi,
>
> is anyone running an HFC based ISDN card with fax receive in parallel to
> Asterisk? Customer needs to start slow, if business succeeds will need
> more than a single HFC based card (probably octobri), but that is in the
> future
On Mon, 2004-11-15 at 22:06, Matt Riddell wrote:
> Jens Hansen wrote:
> > ahh, i see.
> > but that's too much for me, i am not good in php.
>
> Does the error give you a file name and line number?
>
> If so, post the line here and someone will tell you what to change it to.
He's getting the err
I'm using this in production, and it works like a charm. (example from
PHP). Phone numbers have been changed to protect the innocent. ;)
fputs($socket, "Action: Originate\r\n");
fputs($socket, "Channel: Zap/g1d/1234567890\r\n");
fputs($socket, "Exten: 5002\r\n");
fputs($socket, "Priority: 1\r\n")
I am wondering if its possible to play an audio file in a meet me room?
I am also recording the conversations (both sides) of the call. I dont
have to record during the audio play back.
Currently what we have to do is Call the person, start recording, talk
with them, stop recording, transfer t
Im trying to use the auto-dial out in asterisk and having problems
Here is my 1.call file that I move into the outgoing directory:
#
# Create the call on group 2 dial lines and set up
# some re-try timers
#
Channel: Zap/g2/4807480034
MaxRetries: 2
RetryTime: 60
WaitTime: 30
#
# Assuming that y
no, all i got is posted
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt Riddell
Sent: Monday, November 15, 2004 11:06 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Meetme2 - web interface not working
Jen
Jens Hansen wrote:
ahh, i see.
but that's too much for me, i am not good in php.
Does the error give you a file name and line number?
If so, post the line here and someone will tell you what to change it to.
--
Cheers,
Matt Riddell
___
http://www.sineapp
Hi,
is anyone running an HFC based ISDN card with fax receive in parallel to
Asterisk? Customer needs to start slow, if business succeeds will need
more than a single HFC based card (probably octobri), but that is in the
future.
I have an installation with a CAPI ISDN card being shared by
capi
Well I tried just about every combination that I can think of as well as every
combination mentioned and it still doesn't work. Not sure why, maybe it's
just not possible from the Manager API.
Pete
On Monday 15 November 2004 04:56, Peter Svensson wrote:
> On Mon, 15 Nov 2004, Brian West wrote:
ahh, i see.
but that's too much for me, i am not good in php.
thanks anyway
jens
-Original Message-
From: Martin List-Petersen [mailto:[EMAIL PROTECTED]
Sent: Monday, November 15, 2004 11:02 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc: [EMAIL PROTECTED]
Subject: RE
[EMAIL PROTECTED] schrieb:
I suggest you take a look at http://lartc.org/
I already found a mailinglist called this name when "googleing" for
help. But I haven't found that page until now.
There you can find all the info on creating a box for shaping traffic.
Great.
Currently, I configured a P20
On Mon, 2004-11-15 at 21:52, Jens Hansen wrote:
> yes, i have set
>
> define ("DB_TYPE", "postgres"); // mysql or postgres
>
> in defines.php
>
> and i have set
>
> dbtype=postgres
>
> in meetme2.conf.
>
> and the entry is showing up in my postgres db
>
Sorry,
i might not have stated it q
So load asterisk into Slot 1 & 2 and put Commedian in slot 3? Sounds easy
enough. Does that amke voicemail notification work too?
This does seem moderately hokey to have to load it into both slots. I'll
try that shortly with the 2 AdsiProg example you've get below. Is there a
long answer? If
On Mon, 15 Nov 2004, Brian West wrote:
> Ok to cut confusion here
>
> Its:
> Variable: _ALERT_INFO
> Value: somevalue
>
> Its always var/val via manager.
Not in the Originate action it isn't. This is what both the help
show manager command originate
say and what reading the source indicate
yes, i have set
define ("DB_TYPE", "postgres"); // mysql or postgres
in defines.php
and i have set
dbtype=postgres
in meetme2.conf.
and the entry is showing up in my postgres db
-Original Message-
From: Martin List-Petersen [mailto:[EMAIL PROTECTED]
Sent: Monday, November 15, 2004
On Mon, 2004-11-15 at 16:06, Joseph wrote:
> [snip]
> > You are correct. I had it straight in my head but wrote the email wrong
> > :) The parameter I originally meant was "minmessage" which should set
> > the minimum length of the voicemail message in seconds. A quick source
> > code check confirm
On Mon, 2004-11-15 at 16:14, Jerry Geis wrote:
> I does show registered.
>
> HostUsername Refresh State
> sip.broadvoice.com:5060 XXX1184 Registered
This is probably not related but that refresh rate looks funny. Mine has
been 15 since applyin
This is not an Asterisk Problem.
It is a SIP problem and most likely related to the different firewalls the
SIP phones are located behind.
You will need to configure the firewalls, use real addresses for the SIP
phones, or use IAX clients for phones behind firewalls.
Tim.
-Original Messag
Postgresql is correct there :oP.
LIMIT is a mysql specific SQL statement.
--
Slán lait,
Martin List-Petersen
Dublin, Eire
(contact info on --> http://www.marlow.dk/)
On Mon, 2004-11-15 at 21:20, Jens Hansen wrote:
> i checked postgres logfile. it says
>
> ERROR: LIMIT #,# syntax not support
Hi!
I want the SIP-traffic to have the highest priority. I guess the best
method for this is traffic shaping.
I'm using debian with kernel 2.6.5. I installed the tools "tc"
and
"iptables" but I'm not really sure how to use it.
Can anybody help me in providing me a ready-made script?
Thanks!
On Mon, 2004-11-15 at 15:40, Rob Emanuele wrote:
> I got a 390 Adsi phone (unlocked) hooked to my wct400. It seems too work
> pretty well. I'm having two little problems with it.
>
> 1) The idle title screen will not show up unless I manually press service
> and select "Asterix PBX". What seems
Hello:
We have a strange problem with Asterisk 1.0.1
We made a succesfully installation of Asterisk over RedHat. We have a FXO card
installed and configured, and works great.
The Asterisk have a public IP address and the users are behind a firewall and
use invalid IP addresses
We made some test
> > all i have is random echo
> > I have already 4 TA750 with full FXO
> > echocancel=yes and echo training=800
> >
> > - what should i do?
> > - could it be solved with tweaking echo params on *?
> > - is there any additional devices that can be added between Channel
> > Bank and * to get rid off
Jerry:
Check to make sure you have you sip.conf altered and /etc/hosts in the
way outlined by the broadvoice email. I had this same problem. Turns
out that you NOW need a "insercure=very" to receive incoming calls.
Robert
Jerry Geis wrote:
I am still getting a Busy message when I call in
i checked postgres logfile. it says
ERROR: LIMIT #,# syntax not supported.
Use separate LIMIT and OFFSET clauses.
-Original Message-
From: Martin List-Petersen [mailto:[EMAIL PROTECTED]
Sent: Monday, November 15, 2004 3:53 PM
To: Asterisk Users Mailing List - Non-Commercial Disc
Brian West wrote:
Ok to cut confusion here
Its:
Variable: _ALERT_INFO
In CVS head.
In stable, it's still ALERT_INFO.
The same applies to VMXL_URL that is now _VXML_URL in CVS head.
This is due to a change in app_dial where you now are able to
set any variable in the new call leg created by dial() b
I does show registered.
Host Username Refresh State
sip.broadvoice.com:5060 XXX 1184 Registered
On Mon, 2004-11-15 at 15:01, Jerry Geis wrote:
> I am still getting a Busy message when I call in to my broadvoice
> number.
> Is anyone else sti
> all i have is random echo
> I have already 4 TA750 with full FXO
> echocancel=yes and echo training=800
>
> - what should i do?
> - could it be solved with tweaking echo params on *?
> - is there any additional devices that can be added between Channel
> Bank and * to get rid off echo forever?
Ok to cut confusion here
Its:
Variable: _ALERT_INFO
Value: somevalue
Its always var/val via manager.
bkw
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Peter Svensson
> Sent: Monday, November 15, 2004 2:50 PM
> To: Asterisk Users
[snip]
> You are correct. I had it straight in my head but wrote the email wrong
> :) The parameter I originally meant was "minmessage" which should set
> the minimum length of the voicemail message in seconds. A quick source
> code check confirms that any voicemail less than "minmessage" will get
Right, sorry I figured that was what you ultimately wanted to do.
Michael Shuler, C.E.O.
BitWise Communications, Inc. (CLEC) And BitWise Systems, Inc. (ISP)
682 High Point Lane
East Peoria, IL 61611
Office: (217) 585-0357
Cell: (309) 657-6365
Fax: (309) 21
Leandro Morgado wrote:
[EXTENSIVELY SNIPPED]
zaptel.conf:
loadzone=fr
defaultzone=fr
zapata.conf:
---
busydetect=yes
busycount=7
[EXTENSIVELY SNIPPED]
1. Does the Nokia 32 GSM provide you with a hangup tone? (I.E. beep beep
beep once it has hung up)
2. You have specified Fran
I think this has been answered way more then once in the last six months
or so.
> I have already send a message to Digium but I post this again to
> know of anyone who use the G.729 codec.
> My questions are:
> - Do I have to pay 10$ per month or it is only this fee to use the
> codec foreve
Hi,
Works as advised!
Thanks Darly
On 16 Nov 2004 at 8:26, Matt Riddell wrote:
> Darly Coupet wrote:
> > Hi,
> >
> > Thanks for your response. More info as requested:
> >
> > Location: USA
> > FXO connection: Wipphone.com service (similar to Vonage)
> > Analog Telephone Adaptor: Webpho
On Mon, 2004-11-15 at 20:35, Ken Chan wrote:
> Hello,
> Has anyone actually connect a BRI telephone to the
> BRI Card running Asterisk?
Sure Ackermann Euracom P4 and Teles.FON, no problems.
> I have been trying it and no luck so far.
>
> Here is my configuration:
> H/W: T1 Trunk, a few VOIP phon
> I am still getting a Busy message when I call in to my broadvoice number.
> Is anyone else still getting that or found a fix to it?
> I can call out all I want no problem.
>
> This seemed to start happening after the patch was applied.
BV has been working here since the patch was installed last
On Mon, 15 Nov 2004, Peter Osborne wrote:
> I am using the Asterisk Manager API to originate calls and it is working
> well,
> when a call is placed the local phone rings, once you pick it up you can here
> the call ringing the other end. Now, I am using Polycom IP 300 and I have
> them setup
Hi!
I want the SIP-traffic to have the highest priority. I guess the best
method for this is traffic shaping.
I'm using debian with kernel 2.6.5. I installed the tools "tc" and
"iptables" but I'm not really sure how to use it.
Can anybody help me in providing me a ready-made script?
Thanks!
Mic
I got a 390 Adsi phone (unlocked) hooked to my wct400. It seems too work
pretty well. I'm having two little problems with it.
1) The idle title screen will not show up unless I manually press service
and select "Asterix PBX". What seems odd is that if I do not manually
select it it follows the
Other than the standard codec issues? No. disallow=all and allow=ulaw
in [general] in sip.conf. NO other allow= lines.
Chris TenHarmsel wrote:
No one?
On Mon, 15 Nov 2004 12:40:42 -0500, Chris TenHarmsel <[EMAIL PROTECTED]>
wrote:
Hi all,
I've attached the output from asterisk with "set verbo
I have searched a bit on the
Wiki and mailing list archives, but didn’t see direct information
regarding my scenario:
1. Asterisk for
IVR/Voicemail ONLY (no PSTN, no MOH) 2. BudgeTone IP phones and HandyTone 286
ATAs 3. SIP only - separate Proxy+Registrar+CallRouter on other servers 4.
Hello,
Has anyone actually connect a BRI telephone to the
BRI Card running Asterisk?
I have been trying it and no luck so far.
Here is my configuration:
H/W: T1 Trunk, a few VOIP phones, a few analog phones and
a few BRI Phones (Lucent i2021 phone and Tone Commander phone).
S/W: Asterisk 1.0
The $10 is a permanent license.
You only need a license for each *transcoding instance* in the asterisk box.
The Cisco's already have 729 built in. Cisco to Cisco doesn't take any
license, Cisco to voicemail will require 1 license per simultaneous
voicemail connection. Cisco to conference room w
On Mon, 2004-11-15 at 15:01, Jerry Geis wrote:
> I am still getting a Busy message when I call in to my broadvoice
> number.
> Is anyone else still getting that or found a fix to it?
> I can call out all I want no problem.
>
> This seemed to start happening after the patch was applied.
I've appli
No one?
On Mon, 15 Nov 2004 12:40:42 -0500, Chris TenHarmsel <[EMAIL PROTECTED]> wrote:
> Hi all,
> I've attached the output from asterisk with "set verbose 3". During
> the time in the file, I placed two calls with my Zyxel 2000w to a
> Cisco 7912g. The first call worked fine, I was able to ta
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