Hi!
Fabio == Fabio [EMAIL PROTECTED] writes:
Fabio are you using canreinvite=yes on your SIP endpoints definition
Fabio ?
No, I'm using canreinvite=no.
Fabio also check your features.conf, do you have pickupexten = *8 ?
Yes it is:
canopus*CLI show features
Builtin Feature
It's mark on some documentations...
Where do i laucnh qozap ??
Best regards,
Olivier S.
Tzafrir Cohen a écrit :
On Wed, Jun 07, 2006 at 05:44:16PM +0200, Olivier Saulnier wrote:
Hello,
I install the latest release of Asterisk, QuadBri driver.
I compile al; that, copy zaptel.conf file,
Thats just the thing, and it sucks,
because the VoIP implementation actually works very good.
Jon
Fra:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] På vegne af list mail
Sendt: 8. juni 2006 02:34
Til: Asterisk
Users Mailing List - Non-Commercial Discussion
Emne: Re: SV:
On Thu, Jun 08, 2006 at 08:56:52AM +0200, Olivier Saulnier wrote:
It's mark on some documentations...
Where do i laucnh qozap ??
qozap is not a program that you loanch. It is a kernel module that you
load.
Stick the command 'modprobe qozap' somewhere in your init scripts.
Actually, there is
I'm getting this error when compiling:-
make[1]: Entering directory `/usr/src/asterisk.svn/sounds'
--09:22:12--
http://ftp.digium.com/pub/telephony/sounds/releases/asterisk-core-sounds-en-wav-1.4.0.tar.gz
= `asterisk-core-sounds-en-wav-1.4.0.tar.gz'
Resolving ftp.digium.com...
Olivier Saulnier wrote:
It's mark on some documentations...
Where do i laucnh qozap ??
Best regards,
Olivier S.
Tzafrir Cohen a écrit :
On Wed, Jun 07, 2006 at 05:44:16PM +0200, Olivier Saulnier wrote:
Hello,
I install the latest release of Asterisk, QuadBri driver.
I compile al; that,
On Thu, 2006-06-08 at 09:26 +0200, Dave Cotton wrote:
I'm getting this error when compiling:-
make[1]: Entering directory `/usr/src/asterisk.svn/sounds'
--09:22:12--
http://ftp.digium.com/pub/telephony/sounds/releases/asterisk-core-sounds-en-wav-1.4.0.tar.gz
=
Hi,
Can anybody tell me Does Asterisk has a TAPI Interface
sanchal
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Hi,
Can anybody tell me that does asterisk have TAPI interface
sanchal
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check cdr_mysql.conf for userfield=1
turby @ www.canistec.com
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tristan
Sent: Wednesday, June 07, 2006 6:06 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users]
I can set a family/key=value just fine, but how can i delete it?
exten = _200,1,AgentCallbackLogin(||[EMAIL PROTECTED])
exten =
_200,2,Set(DB(AgentsMAP/${CALLERIDNUM})=${AGENTBYCALLERID_${CALLERIDNUM}})
exten = _200,3,Hangup
exten = _201,1,AgentCallbackLogin(||)
exten =
2006/6/8, Paul Hales [EMAIL PROTECTED]:
Another option would be to see if the provider will provide 2 BRI linesthat are tied together in some way.Most of the providers in Australia will do similar things with PRI.PaulH
Do you mean receiving traffic on 2 BRI lines (2 channels spread on 2 separate
good to known.
I played with the idea to buy one of these.
Unacceptably bad voice quality. Point.
You would suggest GrandStream then?
Surely better in my experience.
DV
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The file are there: http://thdei.info/results.zip and http://thdei.info/mos_6_MOS-USA_Test-114_20060605-042551cut-PESQ.png
because, last time I put them in attachment and the
mail was waiting for approvement and I never see it anmore
.
From: Deillon Thomas-WTD008 Sent: 05
June 2006
Hi,
I am looking for a simple php agi script that locates a speeddial
number in a MySQL database and then dials that number.
ie.
exten = 01,1,Noop(speeddial 01)
exten = 01,2,Agi(do database lookup on 01 and finds $NUMBERTODIAL)
exten = 01,3,Goto($NUMBERTODIAL,1)
Anybody know if something
Friends in the community,
I've received many mails saying I'll meet you at Astricon Europe.
The sad answer is no, you will not.
I have nothing to do with Astricon any more. After some arguments,
Steve decided that Astricon, trainings,
the business we had built together - everything
[EMAIL PROTECTED] wrote:
Hi,
Can anybody tell me that does asterisk have TAPI interface
sanchal
No, if you're a windows user, there is asttapi which uses the management
interface though.
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Hi,Installing Asterisk involve tuning NAT and network settings.So, before installing an Asterisk server, I would like to check my network settings.My setup is :
IP Telephony Provider - ISP -- Home router-firewall -- Home LAN --- IPBX and IP Phones
What and how would you check your
Hi all,
I've read a lot of problems with faxing over asterisk. Most of them referred
to Fax over Internet, but I want to connect analog and ISDN fax devices to
asterisk to send and receive faxes over PRI:
+-+ +--+++
| | | || ISDN Fax |
| PRI
I used the configure option '--with-mssql' after freeTDS is installed.
http://uk.php.net/manual/en/ref.mssql.php
Fatal error: Call to undefined function: odbc_connect() in
/var/www/html/odbctest.php on line 3
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Shaun wrote:
I can set a family/key=value just fine, but how can i delete it?
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+DBdel
-- Ben Franklin quote: Those who would give up Essential Liberty to
purchase a little Temporary Safety, deserve neither Liberty nor Safety.
Hi,
Just a little question about digium/sangoma difference of behaviour...
I need to setup 3 E1 connections to 3 different ISDN clock provider ...
Can the TE411P handle this per span or do I have to buy a Sangoma one ?
Thanks in advance !
___
show application DBdel on the CLI. OK this is deprecated but it still
works. Maybe asterisk gives you hints what do use now.
Doug Lytle schrieb:
Shaun wrote:
I can set a family/key=value just fine, but how can i delete it?
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+DBdel
Recent posts indicate people have been having luck with the nokia E60/E7x
phones and asterisk.
I was wondering though if anyone had had any luck with the N80?
I've got the N80 to register with asterisk, and that works just fine.
However, it gives a 486 when I try to place SIP calls to it
Hello List
Ive been trying to use regcontext, but I cant get it to work. Ive setup my sip
peers to have the regexten _[0-9]., so that I can capture all registrations in
a single extension.
But when they register, I can see that the dynamic extension is created, but
none of the rest of the code
8 jun 2006 kl. 11.57 skrev Jon Schøpzinsky:
Hello List
Ive been trying to use regcontext, but I cant get it to work. Ive
setup my sip peers to have the regexten _[0-9]., so that I can
capture all registrations in a single extension.
But when they register, I can see that the dynamic
Hello
Thanks for the answer... Just realized it myself, as your mail arrived :)
Could be a nice feature though.
Jon
-Oprindelig meddelelse-
Fra: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] På vegne af Olle E Johansson
Sendt: 8. juni 2006 12:09
Til: Asterisk Users Mailing List -
I have an issue with DTMF. DTMF is being partly recognised by some
external IVR systems (banks, billing, etc), other IVR systems have
intermittent issues. Call our VSP directly and using their IVR system
without issue, and our internal IVR works just fine. Currently i have
all voip devices
Dear
If I have an extention 111 and 112 on my system but when the user 111 call
the 112 call it through trunk not through local to perform a billing
How can I solve it
Regards
*
No employee or agent is authorized to conclude any binding
We have an asterisk box with the following configuration:
- AMD Athlon XP 2400+
- 512 MB RAM
- SUSE Linux 10.1
- a Digium card TDM400P with 3 FXO
- another Digium card TDM400P with 4 FXS
- asterisk 1.2.7.1
- zaptel 1.2.4
I already checked that those cards aren't sharing interrupts (by cat
Hi;
When connecting via VoipBuster or VoipStunt, I can hear them but they can't
hear me . This happens with VoipBuster or Voipstunt. Registration is done
correctly.
I thought it could be something related to NAT, but I don't have this problem
when using VoipUser or Asterisk2PSTN, for
Christophorus Laube wrote:
show application DBdel on the CLI. OK this is deprecated but it still
works. Maybe asterisk gives you hints what do use now.
As far as I know, dbdel is not depreciated. There is no function for
dbdel yet, at least not that I've read about. dbput and dbget are
Hi all,
I an using MeetMe and I would like to use the -i function to annouce the
join/leave of the user.
However, this require that users record their names. Is there anyway to
remove this?
I just want MeetMe to annouce somethig like A new user has joined the
conference and that need not to
When calling through Plainvoip from my Asterisk at Home box I get the
following log entries.
Jun 8 01:27:25 VERBOSE[5550] logger.c: -- Called plainvoip/1insert #
Jun 8 01:27:26 VERBOSE[2798] logger.c: -- Call accepted by
66.199.240.2 (format g729)
Jun 8 01:27:26 VERBOSE[2798] logger.c:
In article [EMAIL PROTECTED],
Pimjai Wesnarat [EMAIL PROTECTED] wrote:
Hi all,
I an using MeetMe and I would like to use the -i function to annouce the
join/leave of the user.
However, this require that users record their names. Is there anyway to
remove this?
I just want MeetMe to
On 6/7/06, Marnus van Niekerk [EMAIL PROTECTED] wrote:
I have a 1.2.4 * box with two HFC modems using chan_modem_i4l and several
SIP phones and ATA's.
We have a terrible delay on calls between the PSTN (isdn BRI) and the SIP
phones. All internal calls are fine. My first thought was that
Hi all
I have downloaded from openser
and iam trying to integrate voice mail with asterisk
I have read all the docs in the document site
after config, and people recomendation iam able to run the openser successfully
and able to fix the problem calling out side
but when the local user not
Hi,
Is it possible de tell asterisk to increase the volume?
When we place or recieve a call the volume is very low, using a smartphone
or a hardphone.
Thanks for advance
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I've noticed that native music on hold volume seems to be very loud
sometimes. Is there anyway to turn this down? I know when using
mpg123 I can set quietmp3 but what about when using native?
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Turby, Thanks for your replay, but does this mean that * can't play mp3s? I was hoping not to have convert the MOH mp3s that came with * and are supposed to be the native default that * is supposed to be able to play -- not to mention that I don't have sox installed with whick to convert them.
Hi
I want to connect asterisk 1.0.9 ( kernel 2.6.8-2 debian
)with TE110P and Nortel meridian option 11c release 25.40 with NTBK50AA card
which is
An E1 card. But the main problem is the first stage that no sync
occurs the * card never syncs with meridian card
I am
On Thu, Jun 08, 2006 at 12:03:32AM +0100, Marco Mouta wrote:
Marco. i solve this creating adding the meetme extension in the default
context. this extension now is valid for any user.
Hi,
Please check you [general] section in sip.conf
; If you need to answer unauthenticated calls,
If you have to use it, make sure you only use the mpg123 bundled with
the asterisk distribution. mpg123 from any other source (yes, evem
the developer's website) will yield major issues.
On 8-Jun-06, at 8:12 AM, Matt wrote:
I've noticed that native music on hold volume seems to be very
Tristan wrote:
Hi,
Just a little question about digium/sangoma difference of behaviour...
I need to setup 3 E1 connections to 3 different ISDN clock provider ...
Can the TE411P handle this per span or do I have to buy a Sangoma one ?
The digium card has a single on-board clock and you choose
hi all (again). i have this problem. when a people call to meetme and join a conference when this people leave and hangup your phone asterisk can't detect the hangup. all people use analog lines to connect the meetme is any way to tell asterisk to hook when these people leave?
The fact is that I have 2 different E1 (euroisdn) providers and an E1
(euroisdn) connection to a Matra PBX...
The PBX needs to be master and as far as I know the PSTN providers needs
it too...
So I want to be sure that the quad E1 card I'll buy will work without
troubles in this kind of
I'm not using mpg123... I'm using NATIVE MOH!
On 6/8/06, Jason Lixfeld [EMAIL PROTECTED] wrote:
If you have to use it, make sure you only use the mpg123 bundled with
the asterisk distribution. mpg123 from any other source (yes, evem
the developer's website) will yield major issues.
On
Hi, I'm newby here,
reading the handbook and starting playing with *.
What are the audio .gsm files in /var/lib/asterisk/sounds ?
Playback command can only play .gsm ?
how do i convert from .wav to .gsm ?
Thanks a lot
Victor
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Tristan wrote:
The fact is that I have 2 different E1 (euroisdn) providers and an E1
(euroisdn) connection to a Matra PBX...
The PBX needs to be master and as far as I know the PSTN providers needs
it too...
So I want to be sure that the quad E1 card I'll buy will work without
troubles in
Sir
Thanks so much but I have done lots and
lots of googling around and I also had a grip on this file earlier.
I have already tried this but this is for
the T1 scenerio.
I am looking for the ISDN PRI over E1 and
it is not doing any good to me.
The exact card on the Nortel
Hi Victor,
1) you can find sounds.txt file inside asterisk tar file containing the
text of all asterisk sounds and relative filenames
2) Asterisk can play other formats (for example some wav format): search
on wiki
3) for sound conversion see wiki:
Victor Moreno wrote:
Hi, I'm newby here,
reading the handbook and starting playing with *.
What are the audio .gsm files in /var/lib/asterisk/sounds ?
Playback command can only play .gsm ?
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Playback
how do i convert from .wav to .gsm
Oh well. It would have been a nice feature, but with Asterisk's
voicemail-to-email it's not really a necessity.
Thanks for the information!
On 6/7/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
http://www.aredfox.com/eqa.htm#line_10
Check this
Dan
On 08/06/06, Matt Riddell (IT) [EMAIL
Muhammad,
I have been struggling with M1 and * over an E1 for a while myself, but know it's running fine.
Here's my d-channel config:
ADAN DCH 18 CTYP MSDL CARD 08 PORT 1 DES Asterisk1 USR PRI DCHL 8 OTBF 32 PARM RS422 DTE DRAT 64KC CLOK EXT IFC EURO CNTY BEL PINX_CUST 0
ISDN_MCNT 300 CLID
I 'll make some tests with a TE210P and see what happens, I'll post as
soon as I have results...
Asterisk is planned to be at the end of every PRI connection, providing
voip to the PBX and IVR to the customers calling on the 2 E1 lines
connected to Asterisk ...
The MATRA PBX is connected to
Use format_mp3 from asterisk-addons.
It will enable your * to play mp3 without the use of an external process... (if I got it right)
On 6/8/06, Richard Reina [EMAIL PROTECTED] wrote:
Turby,Thanks for your replay, but does this mean that * can't play mp3s? I was hoping not to have convert the MOH
Hi List,
I'm having a problem with detecting incoming dtmf tones with chan_capi,
using an AVM Fritz card. I have set up softdtmf and relaxdtmf, both to 0,
expecting that the capi module will detect the tones, but it did not. I also
set both to 1, expecting that the asterisk dsp functions will
On 06/03/06 22:10 Kevin P. Fleming said the following:
- Michiel van Baak [EMAIL PROTECTED] wrote:
Then the svn automerge thingie Kevin wrote for the asterisk
svn tree is automerging changes to the 'common' tree to all
the server trees.
unrelated to asterisk obviously, but is there
Not sure, may be somebody else can confirm what im going to tell.
From reading the code, it seems the expired timer means the other end
have not recognized the Idle status of your local end (your box). When
you start Asterisk, chan_unicall set the ABCD bits to the unblocked
status and start
I need another fxo line. Has anyone had any experience with connecting
the gsm488 into asterisk?
Thanks,
Jim.
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Greetings all:
In sip.conf, I have configured an entry for Australian VoIP provider
Engin. Sometimes, however, the following error turns up constantly
WARNING: chan_sip.c: Don't know how to indicate condition 9
ERROR: . channel.c: Unable to handle indication 9 for 'SIP/engin-5a0a'
Hi all,
could someone tell me what this does mean bad file descriptor when
trying to start asterisk. It goes till the CLI command and then die with
this message. Below an strace output from asterisk -vc
It's on debian Sarge kernel 2.6.7 with packages from debian VoIP team.
The server
- trixter aka Bret McDanel [EMAIL PROTECTED] wrote:
but they do in 2004 mark said it was one of their biggest revenue
streams. Or do you mean that they dont make any money selling
asterisk
Please post a link (or something) to this quote; selling G.729 licenses has
never been a
Martin Joseph wrote:
On Jun 7, 2006, at 6:55 PM, M.Hockings wrote:
I have a small asterisk setup here with one POTS line, one VOIP SIP
connection an FXS connection to the house phones and a bunch of
softphones. Local calls are routed out through the POTS line and long
distance through the
Do you have the g729 codec?
On 6/8/06, Henry J. Cobb [EMAIL PROTECTED] wrote:
When calling through Plainvoip from my Asterisk at Home box I get the
following log entries.
Jun 8 01:27:25 VERBOSE[5550] logger.c: -- Called plainvoip/1insert #
Jun 8 01:27:26 VERBOSE[2798] logger.c: --
- Matt Florell [EMAIL PROTECTED] wrote:
fixed within a couple weeks and the Digium side being fixed by having
to manually disable the hardware DTMF detection in the wct4xxp.c
driver code every time I upgrade zaptel.
This is no longer needed (editing the source); there is a module
- Steve Underwood [EMAIL PROTECTED] wrote:
other DSP functions for telecoms. What makes you think these are
foundry
chips?
They are (were). They are now being manufactured at a different facility.
--
Kevin P. Fleming
Senior Software Engineer
Digium, Inc.
- Matt Riddell (IT) [EMAIL PROTECTED] wrote:
What does the onboard DSP do when used with Asterisk? Did Digium or
someone put code inside Asterisk to hand off the
processing/transcoding
to a Sangoma card?
According the Sangoma data sheet, the Octasic part _is_ the DSP (which it is,
in a
In a dual server configuration one of the two servers fail with:
WARNING[3705]: res_crypto.c:335 __ast_sign_bin: RSA Signature (key
sintel-voip) failed
NOTICE[3705]: chan_iax2.c:5187 authenticate: Unable to sign challenge
withy key
WARNING[3705]: chan_iax2.c:7191 socket_read: I don't know how
- Dave Cotton [EMAIL PROTECTED] wrote:
Update after another look it isn't, there is only a gsm version.
That is correct; the Spanish sounds and the non-GSM sounds will not be
available until Asterisk 1.4 is released.
--
Kevin P. Fleming
Senior Software Engineer
Digium, Inc.
Hi,
I have a custom agi which at times does not exit gracefull and
crashes in between. The logging options are set to the maximum but I
dont see something conclusive in the asterisk log.
I have noticed it crash after issuing the SAY NUMBER and GET DATA
agi commands and the agi is spawned with no
- Matt [EMAIL PROTECTED] wrote:
I'm not using mpg123... I'm using NATIVE MOH!
No, the native file playback method does not offer any means to manipulate the
volume of the sound being played. If you need to, you can edit the MOH files
themselves using your tool of choice (sox, Audacity,
Hello,
I have to dial prefix 9 for non local numbers however
when i missed calls i Can't redial this number
because of 9 is not append .
I use polycom phones .
What Can i do ?
Harry
__
Do You Yahoo!?
En finir avec le spam? Yahoo! Mail vous
Jason Lixfeld wrote:
If you have to use it, make sure you only use the mpg123 bundled with
the asterisk distribution. mpg123 from any other source (yes, evem the
developer's website) will yield major issues.
mpg123 is NOT bundled with Asteirsk. make mpg123 will DOWNLOAD the
mpg123 source
On Thu, 8 Jun 2006, Esteban Guana-Jarrin wrote:
Hi List,
I'm having a problem with detecting incoming dtmf tones with chan_capi, using
an AVM Fritz card. I have set up softdtmf and relaxdtmf, both to 0, expecting
that the capi module will detect the tones, but it did not. I also set both to
Hi All,
I have setup [EMAIL PROTECTED] 2.8 and using Digium TDM400P cards
Whenever I dial out and finish the conversation and put the SIP Snom320
phone down, it rings back twice!!!
If you pick up the phone there is no answer.although you think it's a
genuine call!!
If I change the
Tristan wrote:
I 'll make some tests with a TE210P and see what happens, I'll post as
soon as I have results...
Asterisk is planned to be at the end of every PRI connection, providing
voip to the PBX and IVR to the customers calling on the 2 E1 lines
connected to Asterisk ...
The MATRA PBX
I have installed 1.2.9.1 and it has no /var/spool/asterisk/outgoing
directory. I must have missed some change in this addition when upgrading. Does
anyone know where the automatic outgoing call directory has gone?
Jordan Novak
Communications Technician
In a dual server configuration one of the two servers fail with:
WARNING[3705]: res_crypto.c:335 __ast_sign_bin: RSA Signature (key
sintel-voip) failed
NOTICE[3705]: chan_iax2.c:5187 authenticate: Unable to sign challenge
withy key
WARNING[3705]: chan_iax2.c:7191 socket_read: I don't know how to
Peter,
Perhaps you have not followed the thread over the last few days about
DTMF feedthru??? Here is what I sent out to another list kind of summing
it up
Regarding DTMF pass thru problems when using the SPA-3000 and *. The
problem manifests itself as the inability to pass DTMF over the
Do you have the g729 codec?
On 6/8/06, Henry J. Cobb [EMAIL PROTECTED] wrote:
Jun 8 01:27:26 VERBOSE[2798] logger.c: -- Format for call is g729
...
Jun 8 01:27:29 DEBUG[2798] chan_iax2.c: Ooh, voice format changed to
256
Yes, and that works fine when talking with the phone itself, as
2006/6/8, Kevin P. Fleming [EMAIL PROTECTED]:
And yes, when Digium's Octasic-based module starts shipping (currently in beta testing),Could you elaborate ?Any schedule ?Cheers
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Harry,You can use the prefix in your dial string instead of actually dialing it. Dial(Zap/g0/9${EXTEN})AlexOn 6/8/06,
[EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Hello,I have to dial prefix 9 for non local numbers howeverwhen i missed callsi Can't redial this numberbecause of 9 is not append .I
On 6/8/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
I have to dial prefix 9 for non local numbers however
when i missed calls i Can't redial this number
because of 9 is not append .
I use polycom phones .
What Can i do ?
My preferred answer to this question is to not use a '9' prefix.
Kevin P. Fleming wrote:
- Matt Riddell (IT) [EMAIL PROTECTED] wrote:
What does the onboard DSP do when used with Asterisk? Did Digium or
someone put code inside Asterisk to hand off the
processing/transcoding
to a Sangoma card?
According the Sangoma data sheet, the Octasic part
Is the 94x any better? seems without backlighting, any are
next to useless.
The SPA-9x2 have backlit displays.
Nabeel
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On Thu, 2006-06-08 at 16:28 +0200, [EMAIL PROTECTED] wrote:
Hello,
I have to dial prefix 9 for non local numbers however
when i missed calls i Can't redial this number
because of 9 is not append .
I use polycom phones .
What Can i do ?
RTFM?
--
Dave Cotton [EMAIL PROTECTED]
Hi all,We are running Asterisk 1.2.7.1 on our Dell Poweredge 2850 and are having massive sound quality issues.We are experiencing call quality issues for our
remote location, namely calls cutting out and breaking up for our
agents. The two main issues seem to be 'popping' and 'dropping' - popping
Kevin P. Fleming wrote:
- Matt [EMAIL PROTECTED] wrote:
I'm not using mpg123... I'm using NATIVE MOH!
No, the native file playback method does not offer any means to
manipulate the volume of the sound being played. If you need to, you
can edit the MOH files themselves using your tool of
Is there any setting
in the voicemail that will send the voicemail file in a type that is recognized
on a Blackberry?
Kerry GarrisonDirector of
Technical ServicesTech Data Pros - Orange County's Mobile IT Service
Provider(949)502-7819 x200- [EMAIL PROTECTED]http://www.techdatapros.com
Sorry about stupid question but I would liek to get
help about Zap channel.
We would like to get early media on session in
progress from zap channel.
But using the standard exten =
_X.,1,Dial(Zap/g1/${EXTEN}|60|o) I don't hear any audio until someone pickup the
phone.
Now I can't now if
Mike Fedyk wrote:
I have heard good things about the D-Link DES-1226G switch ($150 at
newegg). If you can run a separate cable to the computer and phone. If
you can't run the extra cables, then configure your phone to tag itself
as part of the voip vlan and let the switch tag everything else
Figuring I knew what I was doing (I didn't - surprise) I added a
totally unnecessary line in /etc/asterisk/extensions_additional.conf a
couple of days ago. Troubleshooting a dialing rule issue, I'm now
realizing that FreePBX is updating its database with the new settings
but is not
On 6/8/06, whois wes [EMAIL PROTECTED] wrote:
I can call in from either a Zap or SIP channel and have
sound quality issues, so the network is probably not causing the issue - a
purely Zap channel still experiences pops and drops. Same with a purely SIP
channel. The recorded call doesn't seem
Try
the 'g' option in your dial statement:
exten =
_X.,1,Dial(Zap/g1/${EXTEN}|60|og)
hth
-Original Message-From: Rosario Pingaro
[mailto:[EMAIL PROTECTED]Sent: Thursday, June 08, 2006 10:00
AMTo: asterisk-users@lists.digium.comSubject:
[Asterisk-Users] early session
hello
how can i configure asterisk to use soft sip phone
and when asterisk is running how can I know he work correctly
thanks
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- Olivier Krief [EMAIL PROTECTED] wrote:
Could you elaborate ?
Any schedule ?
No, there is nothing really to elaborate... and this is not a commercial
mailing list, so I'm not comfortable talking about it more here anyway :-)
If you need more details, contact our sales department.
--
- Mike Fedyk [EMAIL PROTECTED] wrote:
Will it have a 1024 tap echo can on all 96 channels? What about 8 T1
support like sangoma?
Those are completely unrelated questions; there is no need for an 8-span echo
can module when there is no 8-span T1 card :-)
It uses the identical Octasic
STDERR from your agi will be shown on
asterisks tty. If youre using safe-asterisk to start, I believe
this is redirected to tty9 Or, if you can afford to take asterisk down
momentarily, you could just start asterisk without backgrounding it and youll
see what your script has to say there.
I couldn't find one but I didn't look too hard.
To be honest, the Blackberry is so easy to use with one hand I dropped the
issue.
We actually switched to Windows Mobile devices which suck compared to the
Blackberry for email/ease of use but I can now one click listen to my voicemail
without
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