Hi I really hope somebody can help me out.
I have an asterisk installation working on a Redhat 9
system. I now want to add the MySQL functionally to it. However when I make the
necessary changes, (downloading the add-ons, and changing the Make file) the
make fails.
I have looked
Hi.
I'm not being able to make my Voicetronix Openswitch 12 work with
Asterisk. When making a call from a SIP-phone to the PSTN, the SIP-phone is
ringing, the Asterisk says that it is ringing, but the phone I'm ringing is
not ringing. I've seen in the mail list that other people have had
Umar Sear wrote:
Hi I really hope somebody can help me out.
I have an asterisk installation working on a Redhat 9 system. I now
want to add the MySQL functionally to it. However when I make the
necessary changes, (downloading the add-ons, and changing the Make
file) the make fails.
I
Looking for a shining star in c# and or c++ .net development to take the
reins on an Asterisk PBX management interface. The customer requests
delivery of a working prototype by 9 April 2004, so time is of the essence.
This is a paid contract and is open to a developer anywhere in the world.
In article [EMAIL PROTECTED],
Umar Sear [EMAIL PROTECTED] wrote:
I have an asterisk installation working on a Redhat 9 system. I now want
to add the MySQL functionally to it. However when I make the necessary
changes, (downloading the add-ons, and changing the Make file) the make
fails.
I
772 is generally what cell phone companies in florida use. Nextel,
Sprint, att, etc.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tim Sailer
Sent: Wednesday, March 17, 2004 10:21 PM
To: Asterisk Users
Subject: Re: [Asterisk-Users] local VoIP in
That's not actually correct. 772 area code is Port Saint Lucie, Fort Pierce,
Vero Beach, Stuart, and Jensen Beach. That's Martin, Saint Lucie, and Indian
River counties east coast of Florida just north of West Palm Beach.
-Bill
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
There are software solutions you could refer to, software based, or hardware based.
Look at www.voip-info.org for both solutions available.
To give you a hint;
For Software based solution, you can try various managers available for asterisk
(Windows and Linux based) in order to get line status.
I have three * servers that are inter-connected, registering with each
other. Up until yesterday I was authenticating all three with MD5, and
all was working fine.
Yesterday I switched to RSA, and everything is working as well. I can
see AUTHENTICATED messages on the console if one of the
Hi all,
It seems this week's release of spandsp fixed the major problems in the
previous release, but still people have had a lot of trouble. Working
with some of those who tried the software and gave me good feedback, I
have identified some apparently common bugs in fax machines, and I have
Haha,
Well why not? Everyone has to eat at the end of the day!
Is it worth considering setting up an asterisk-trading mailing list specifically for
this purpose?
Hotlinks Internet Services offers Voip grade bandwidth on our Juniper powered network
and colocation space in the Major London
Well why not? Everyone has to eat at the end of the day!
Is it worth considering setting up an asterisk-trading mailing list
specifically for this purpose?
But surely we'd all just end up trying to sell to each other that way! At
least being on the main mailling list means that we have plenty
Linus Surguy wrote:
Well why not? Everyone has to eat at the end of the day!
Is it worth considering setting up an asterisk-trading mailing list
specifically for this purpose?
But surely we'd all just end up trying to sell to each other that
way! At least being on the main mailling list
There is a configuration and billing system for *. Refer to www.vidanetwork.com
Do you Yahoo!?
Yahoo! Mail - More reliable, more storage, less spam
Linus Surguy wrote:
ob: Magrathea offers A-Z IAX termination, origination blah blah blah
blah.
I asked a while ago, and you passed me to a reseller who never answered
my question - How much to terminate a call in the UK?
David
--
David Coulsonemail:
From: ×áëâáôæüãëïõ Êþóôáò [EMAIL PROTECTED]
Reply-To: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] RE: line status
Date: Thu, 18 Mar 2004 15:09:52 +0200
There are software solutions you could refer to, software based, or
hardware based.
Look at www.voip-info.org for both
HEY! I'm doing research and testing for my Thesis on a prototype SIP PBX for a
facility of 20-30 users. (T100P / Atlas 550series / Cisco Routers switches)
A couple of concerns that have come up are:
1. Has there been any known issues concerning asterisk with the new 64-bit
processors?
2.
How can I get the latest CVS of chan_sccp
The way described on Zozos webpage seems not to work:
[EMAIL PROTECTED]:~ export CVSROOT=:pserver:[EMAIL PROTECTED]:/var/lib/cvs/
[EMAIL PROTECTED]:~ cvs login
(Logging in to [EMAIL PROTECTED])
CVS password:
/var/lib/cvs/: no such repository
cvs [login
But surely we'd all just end up trying to sell to each other that way! At
least being on the main mailling list means that we have plenty of customers
to prey on?!
It depends which way you look at it, it could just be more people to give you hassle
and waste your time!
I guess I was
I'm using some even older 32 ms 2551 and 2531's on my fxo and fxs lines.
They work just like TC says. No training time, the echo is just gone.
There is a serial, menu-driven interface on the Tellabs racks that makes
them really easy to configure. My only complaint is that the rack is
designed
Hi Steve.
On Thu, 18 Mar 2004 22:06:46 +0800
Steve Underwood [EMAIL PROTECTED] wrote:
snip
There is now a new tarball at
ftp://ftp.opencall.org/pub/spandsp/spandsp-0.0.1b.tar.gz Please try
this, and report any problems you find. This version has the following
changes:
A
Hi,
The messages produced by asterisk console, in vvv mode,
what are the numbers after the brackets?
in this example, /4 and /5
= Releasing [EMAIL PROTECTED]/4 and IAX2[ulf]/5
Are these session numbers or?
Are they reused?
When the first call comes after asterisk is restarted, they begin
Hi
everyone,
Is there an
application in Asterisk which can be used as a fax receiver?
something
like:
exten =
1234,1,ReceiveFax(...)
exten =
1234,2,ForwardReceivedFax( emailaddress )
Tomica
Hi,
You can build a solution with spandsp library. You will need an email
server too.
http://www.opencall.org/instruction
Daniel
Tomica Crnek wrote:
Hi
everyone,
Is
there an application in Asterisk which can be used as a fax receiver?
something
like:
exten
=
Search on DTMF
- Original Message -
From: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, March 17, 2004 2:57 PM
Subject: Re: [Asterisk-Users] can't logon to voice mail - bad password
Paul,
Do your other extensions work?
If you have only one extension, note that the filename
Paul,
What Client are you using? Also what is the output on the console when you
dial into the voicemail from that extension?
I had the same issue using a BT100 set to early dial. It turned out to be a
DTMF issue. Once I played with different DTMF options both on the phone and
in the * configs,
Hi,
I'm confused about a config we have going where there is
NAT router -- 192.168.1.101 linux+asterisk
PC -- 192.168.1.104 WinXP with X-Lite
At another location:
NAT Router -- PC X-Lite xxx.xxx.xxx.xxx
The remote works fine with *, can use the FXO line, can call FWD members
thru *,
Hi !
I need a quick help configuring an Wildcard E100P ...
Inbound calls are working ok, but I can not call out, dialing 20 only
gets me a line dial tone, but no call is made; same stuff with _0.
direct dialing
Please provide some suggestions if you have ! TNX !
This is my actual config
It's just a short cut for Asterisk! In stead of spelling out the
Asterisk PBX most just type *.
- Original Message -
From: Eric Kirkland
To: [EMAIL PROTECTED]
Sent: Thursday, March 18, 2004 10:36 AM
Subject: RE: [Asterisk-Users] * and PrePaid
Ok, Im definitely too geeky. Im new to
On Thu, 2004-03-18 at 07:47, Carey Jung wrote:
Anybody have a list of area codes and prefixes for which Nufone can provide
DIDs? I can't find any such list on their site.
Michigan only, but I believe they have decent coverage within Michigan.
I seem to recall they were planning on Chicago
Check the extensive thread regarding this EXACT ISSUE in the mailing
list archives.
On Thu, 2004-03-18 at 04:36, tim mickelson wrote:
Hi.
I'm not being able to make my Voicetronix Openswitch 12 work with
Asterisk. When making a call from a SIP-phone to the PSTN, the SIP-phone is
Hi,
I am new to VOIP and Asterisk. I have downloaded and
installed Asterisk in my Linux machine and tested
using asterisk c command it works fine. It's
an excellent product.
Without using any of Digium's hardware or T1 or E1
interfaces
, can i do voice chat between two computers
Hello,
I have H323 up and running. However, I do not see the T38
codec as an option. I have looked through the mailing-list and saw a couple of
postings with T38 listed in the codec list for the oh323.conf file. Am I
missing something here?
Regards,
Mark
I am using NuFone H323 module.
Following on extensions.conf works (x.x.x.x = is the IP address)
extensions.conf
---
exten = 2000,1,Dial(H323/[EMAIL PROTECTED])
Following do not seems to work, but I need to dial out using following,
due to various reasons. Why I cannot dial out
On 18/03/04 15:40, Kevin wrote:
I seem to be having problems using my sound card with asterisk and
gnophone in a Gentoo system (not sure if it being Gentoo is important
or not, but thought I'd mention it just in case). I have the following
errors when starting gnophone:
Looks to me like
Hi,
-Original Message-
Some fax machines send a little less than the specified
1.5 seconds of training test data, so the training test
failed every time. I now only look for 1.25seconds of
training test data.
I think this is still on the long side? I have a few fax-services
I just pushed out a snapshot of the -devel version of monastery.
ftp://ftp.buoy.com/pub/asterisk/monastery-devel.tgz
Tim
--
Tim Sailer Coastal Internet, Inc.
Network and Systems Operations PO Box 726
http://www.buoy.com
Hello, All.
I have a bit of a peculiar problem with MWI.
First some basics:
Hardware: PT390 connected to a Digium TDM400P
I included mailbox=100 statement in zapdata.conf and the mailbox is
defined in voicemail.conf.
Everything works OK. I receive VMs, and when I pick up the handset I get
the
This was posted last year by Mark. I figured I'd repost it to refresh
peoples memories.
Please stop posting commercial postings and announcements to the *-users
and *-dev. Let's self moderate so the list doesn't have to be moderated
James
-- Forwarded message --
Date: Mon, 7
Sure thing. You're going to have to get SIP involved though. This
means using sip.conf to create new sip users.
Do a search on www.voip-info.org for sip.conf and it will explain how to
configure a user for SIP.
Then you'll need SIP clients (hard VoIP phones, or SIP soft clients such
as Windows
I don't think it can do these things. Yes I know the web pages says so but
the book doesn't and neither does Yan, Pulvers techy.
Mark
Steven Thomas said:
Hi,
I have received my WiSIP phone - works well for basic functions of call
answer and hang-up!
Does anyone know how to enable Dual
So give us a commercial list.
Please :)
Panny
- Original Message -
From: James Golovich [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, March 18, 2004 5:10 PM
Subject: [Asterisk-Users] Should List be Moderated?
This was posted last year by Mark. I figured I'd repost it to
I just tried this, and it's not working for me.. I can't call a 2600 or a
CCM... What version of OpenH323 and PWLIB did you all use?
- Original Message -
From: Marian Durkovic [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, March 18, 2004 10:35 AM
Subject: [Asterisk-Users]
Hi, I'm testing Nikotel with Asterisk.
Sound quality is Ok, but I can´t manage to have a call longer then 1 minute
After 1 minute or so, my * exchanges some SIP messages with Nikotel and the
call ends with maximum retries error.
Debugging the SIP messages, I see 2 IP´s in the VIA header, the
It is from this extensive thread that I fond that I should put a comma in the
dial string, that didn't help, now what should I do? This thread regarding this
issue does not help me.
tim
Check the extensive thread regarding this EXACT ISSUE in the mailing
list archives.
On Thu, 2004-03-18
And thus Asterisk-Biz was born.
(http://lists.digium.com/mailman/listinfo/asterisk-biz)
On Thu, 18 Mar 2004, Panny Malialis wrote:
So give us a commercial list.
Please :)
Panny
- Original Message -
From: James Golovich [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent:
... just installed this. The database updates OK but status.php shows no
active channels (either SIP to SIP or SIP to voicemail).
Iain
--On Thursday, March 18, 2004 11:51 am -0500 Tim Sailer [EMAIL PROTECTED]
wrote:
I just pushed out a snapshot of the -devel version of monastery.
Hi all,
in an effort to create a SIP - H.323 translator we've found and fixed
several problems in H.323 channel. These inlcude:
for SIP-H.323 calls
- no ringback tone
- ringback not related to H.323 events
- one-way audio with Cisco CallManager
- incorrect Caller ID
for
Hi,
Could you tell me what role the ASTERIX can play.
Is it Sip Registry Server ?.
Could it work as Proxy Server ?
Thanks
Ahmet
BerliKomm Telekommunikationsgesellschaft mbH
Ahmet Balamir
TC,
Appreciated your help and will try out TelLabs card and see if we can get
rid of echo.
Yesterday I did some changes in TX and RX attenuation setting on Channel
bank and it reduces the echo, but it is not yet vanished as we wanted.
Any way Thanks.
Regards,
KD
Date: Wed, 17 Mar 2004
Title: RE: [Asterisk-Users] Softfax/spandsp
Hi,
This week's spandsp release is a big step ahead, there are no problems
so far receiving faxes from our HP OfficeJet R80xi and Panafax UF-560.
Still, we cannot get anything from our Dialogic fax boards. It goes like this:
-- Executing
Now if we can just get the list software configured to bounce untrimmed
posts with multiple copies of the footer, we would be all set!!
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
quote who=James Golovich
And thus Asterisk-Biz was born.
(http://lists.digium.com/mailman/listinfo/asterisk-biz)
[EMAIL PROTECTED]: unknown user: asterisk-biz-request
So, when will it be fully up?
--
END OF LINE
-MCP
___
Asterisk-Users
Will these be available on the CVS? Devel or Stable?
Hi all,
in an effort to create a SIP - H.323 translator we've found and
fixed
several problems in H.323 channel. These inlcude:
for SIP-H.323 calls
- no ringback tone
- ringback not related to H.323 events
- one-way audio with
On Thursday 18 March 2004 11:29, Alastair Maw wrote:
On 18/03/04 15:40, Kevin wrote:
I seem to be having problems using my sound card with asterisk and
gnophone in a Gentoo system (not sure if it being Gentoo is
important or not, but thought I'd mention it just in case). I have
the
Rob,
Thanks for the info. Since it seems like BRI is not too popular in the
U.S., I think that I will try to pick up a DIVA PCI and see if it will
work with CAPI or i4l.
-Tor Roberts
Rob Fugina wrote:
On Wed, Mar 17, 2004 at 12:25:49PM -0800, Tor Roberts wrote:
Hi all,
I have been using
Is there any reason the only country you can choose is USA? There are more
countries than that...
:-)
Matt
I thought thats what http://www.iaxtel.org was all about.. :)
___
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[EMAIL PROTECTED]
On 2004 Mar 18, at 11:29, tim_mickelson wrote:
It is from this extensive thread that I fond that I should put a comma
in the
dial string, that didn't help, now what should I do? This thread
regarding this
issue does not help me.
You cannot put either the , or the directly into a Dial string.
From my extensions.conf:
[globals]
#TRUNK=Zap/1
VPBPAUSE=,
[trunkld]
exten = _91NX,2,Dial(vpb/1-1/${VPBPAUSE}${EXTEN:1})
exten = _91NX,1,Dial(vpb/1-2/${VPBPAUSE}${EXTEN:1})
On Thu, 2004-03-18 at 11:29, tim_mickelson wrote:
It is from this extensive thread that I fond that I
Eric Wieling wrote:
Michigan only, but I believe they have decent coverage within Michigan.
I seem to recall they were planning on Chicago DIDs, but I don't know
the status of that.
Jeremy from Nufone and Mark Spencer were both at this week's WISPCON in
Chicago. From the smell of it we won't
I'm investigating asterisk to use as a replacement for an aging Lucent
PBX in our district office, as well as replacing the Centrex/intercom
based systems at our schools.
I'm curious if any other schools/districts are using asterisk? If so,
I'd certainly be interested in talking about your
Yesterday I tried to connect an * server with an X100P card to an
extension of an ATT PBX. The X100P never could detect the line and always
gave an alarm. Is there some special type of config that must be done to
connect an FXO port to an extension of a PBX?
--
Carlos Chavez
Corporativo
On Thursday 18 March 2004 13:46, Kevin wrote:
On Thursday 18 March 2004 11:29, Alastair Maw wrote:
On 18/03/04 15:40, Kevin wrote:
I seem to be having problems using my sound card with asterisk
and gnophone in a Gentoo system (not sure if it being Gentoo is
important or not, but
Carlos Chavez wrote:
Yesterday I tried to connect an * server with an X100P card to an
extension of an ATT PBX. The X100P never could detect the line and always
gave an alarm. Is there some special type of config that must be done to
connect an FXO port to an extension of a PBX?
I had that
We are using Asterisk in K12 with similar goals. We have pretty much decided to
go with it district wide. It will be a 200-400 phone installation across 9
sites. We also looking at eliminating a Centrex system (called Plexar in our
area). Current status is that I have patched the * in front of our
Hello Ahmet,
Asterisk is more than a proxy. Its an entire PBX. At a basic level it can
be used as a proxy though.
[EMAIL PROTECTED] said:
Hi,
Could you tell me what role the ASTERIX can play.
Is it Sip Registry Server ?.
Could it work as Proxy Server ?
Thanks
Ahmet
On 18/03/04 18:46, Kevin wrote:
Thanks for your reply, Alastair. I did use that guide in getting myself
set-up with sound, and do have alsa-oss installed:
You need to have it all insmod'ed as well (which I guess it will be):
[EMAIL PROTECTED] almaw # lsmod | grep oss
snd-seq-oss
Mark Phillips wrote:
[EMAIL PROTECTED] said:
Is it Sip Registry Server ?.
Could it work as Proxy Server ?
Hello Ahmet,
Asterisk is more than a proxy. Its an entire PBX. At a basic level it can
be used as a proxy though.
My favourite subject... :-)
No, Asterisk is not even close to a SIP
On Thu, 18 Mar 2004, Chris Hobbs wrote:
I'm investigating asterisk to use as a replacement for an aging Lucent
PBX in our district office, as well as replacing the Centrex/intercom
based systems at our schools.
I'm curious if any other schools/districts are using asterisk? If so,
I'd certainly
Hey all. We've just recently purchased a T100P in order to provide VoIP
to a remote office. We've interfaced it with a DS1-formatter on our
Mitel GX5000 switch. I realize that plugging the * PBX into this class
5 switch isn't the best situation to have in the world...but hey it's
what we've
TC,
Thanks for your recommendation. Looking at sourcing one now. This is
great news.
As I understand it, you need the card, Chasis, and Power Module, and we
should be up and running?
Thanks,
Brent
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL
Dear Steve
sorry for my last bug report (not full reported well) I have now reinstall
your spandsp 1b but i have this type of error...
-
Test Fax station Canon B150
Something you should look at in a school setting is the Wisip wireless
802.11b phone from Pulver. We are looking at this to help reduce cellular
costs and also as a possible teacher phone.
Not unless transfers aren't important to you. I have been unable to
determine how to initiate a
I'll answer my own question ...
If you don't call the database asterisl you need to edit in the name you
do use to status.php otherwise monastery behaves as though nothing is
happening rather than flagging an error ;-)
Iain
--On Thursday, March 18, 2004 5:51 pm + Iain Stevenson
[EMAIL
On Thu, 2004-03-18 at 15:38, Andrew Kohlsmith wrote:
snip
Can anyone explain why the hell it seems impossible to find a VOIP phone
with a good selection of NORMAL (5 years ago) cell-style ringtones? Not
tunes, not a POTS ringback tone, but a good selection of shrill, easy to
identify
On Thursday 18 March 2004 14:57, Alastair Maw wrote:
On 18/03/04 18:46, Kevin wrote:
Thanks for your reply, Alastair. I did use that guide in getting
myself set-up with sound, and do have alsa-oss installed:
You need to have it all insmod'ed as well (which I guess it will be):
[EMAIL
Hi,
I have a partial working installation with zaphfc.
Incoming call :
For incoming call, seems work fine. But the sound is very bad with bounce
short crashing sound. Same sound with echo cancel off or on.
SDA work fine.
Another problem, it's seems that's zaphfc don't reset correctly the line.
I've got my GnuGK box listening on 1720.. CCM thinks it's an OH323
gateway with 245 tunneling... * is registering the # as being an
extension in GnuGK.. I call the # and I see the port 1720 light up with
tcpdump.. but gk -ttt isn't showing my anything and nothing gets send to
the * box... What am
Anyone know if Grandstream ever plan to implement another tone on the
BT-101? To me, it's very weird hearing ringback as the ring-in sound.
Cheers,
Chris.
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
Can anyone explain why the hell it seems impossible to find
a VOIP phone with a good selection of NORMAL (5 years ago)
cell-style ringtones? Not tunes, not a POTS ringback tone,
but a good selection of shrill, easy to identify and
businesslike ringing tones???!?!?!??
No kidding. Imagine my
There is no mention of * there at all or maybe I am blind :(
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Frank Norman
Sent: Friday, 19 March 2004 1:32 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] * and PrePaid
There is a configuration and
All,
I have my AGI working. I am placing a call in the outgoing directory
and running my AGI. Once the call is places and answered I
then need to send DTMF tones. Like 101.
How can I do this in the AGI? I did not see and commands for it.
Thanks,
Jerry
Title: RE: [Asterisk-Users] Softfax/spandsp
Hi,
We've tried J2 faxing with the newest release of spandsp and found
the same issue as with our own Dialogic based faxing.
Steve, we can fax to your system from our platform and/or J2 if you
think it can help.
Here is what we see on incoming
Can anyone decipher these error messages?
Mar 18 18:10:21 WARNING[131081]: chan_zap.c:5949 zt_pri_error: PRI: Read on
39 failed: Unknown error 500
Mar 18 18:10:21 NOTICE[131081]: chan_zap.c:6664 pri_dchannel: PRI got event:
6 on span 1
Thanks, TL
Title: RE: [Asterisk-Users] Softfax/spandsp - page cut-off
Hi,
Faxing from Dialogic and J2 is one problem. Another problem is a page
cut-off - happened 2 times out of 7-8 when faxing from the real fax machine.
...
Get at 9600
Changed from phase 3 to 5
Fast carrier up
Coarse carrier
Hi Folks,
Anyone having issues with FWD lateley? It seems that ever since they sent
me a notification about my voicemail I've been unable to sucessfully make
calls to my WA phone number which is forwarded to FWD.
Also, on my office machine I'm unable to properly register with FWD. I get
a lot of
What sound chip are you using? I thought I had the via82xx and spent a
couple days jacking with it before I figured out I was wrong.
Here's my alsa setup in modules.conf:
# --- ALSACONF verion 1.0.0pre1 ---
alias char-major-116 snd
alias char-major-14 soundcore
alias char-major-15 off
Michael,
As far as I'm aware, RedHat 9 uses the ALSA sound drivers.
you need to prevent asterisk from loading the OSS channel driver with:
noload = chan_oss.so
in your modules.conf
-Ben
--
Computer games do not affect kids!
If Pac-Man had effected us as kids then we would now be running
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To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
From what I know is that if you have a private IP on your asterisk box, and
only a private IP, your box will send out SIP messages containing your
private IP in the FROM field.
try to add this in your sip.conf
externip=63.88.139.198
David
- Original Message -
From: Mark Phillips [EMAIL
Hi to everyone
When I dial a phone numer using my IConnectHere acount I get this message.
Can someone tell me what it is?
Mar 18 18:59:37 WARNING[1217602880]: dsp.c:1424 ast_dsp_process: Unable to
process inband DTMF on 1 frames
Mar 18 18:59:37 WARNING[1217602880]: dsp.c:1424 ast_dsp_process:
Out of everyone using the 7960 currently, what would you say is the best
firmware to use w/ asterisk?
What's the most compatible / stable?
In addition, is there a better / easier or straightforward tutorial to
upgrading the firmware?
Thanks in advance
kindly tell me what is difference b/w loopstart, kewlstart, groundstart for FXO or FXS
devices
Thank you
--
Atif Rasheed
Convergence (Business Systems)
http://www.convergence.com.pk
--
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[EMAIL PROTECTED]
I want to set up a Linux answering machine/voice mail deal that will
e-mail me phone messages. I looked into the voice modem (vgetty)
stuff, but I'm not getting a warm fuzzy feeling about it from reading
the mailing lists - much of it old and not encouraging - and it seems
that there is not a
I kinda like it .. ;)
Nice conservative.
OTOH, the new snom 200 I just got today has some reeeaaally
weird ring tones (and nothing really 'traditional').
Now, maybe we should take a lesson from the cell-phone
people, and talk manufacturers into letting us download
ringtone(s).
Cheers,
WW
Thought I'd drop a note here in case anyone else has been experiencing
this. Cisco routers running NAT are liable to crash when doing NAT
translation of SIP packets. Symptoms are a watchdog caused software
abort in the IP Input process. The router then reloads. In my case
this has been
Chris,
I read your message below, posted to the Asterisk-Users board. I don't
represent or currently have a customer with an asterisk installation in a
school district environment, but if you have any questions, I'd be happy to
chat with you. I'm a network and asterisk private consultant.
Ack, sorry for the reply to the message board on my last reply to this
topic, I forgot to replace the To: person.
Thanks for the replies in advance telling me that that wasn't on topic, I
consider myself in error, and offer the appropriate apology.
-Chris
I ran the up2date and installed newest kernel 2.4.20-30.9 and rebooted the
modules for the zaptel drivers wcfxo and wcfxs didn't load I reboot with
kernel 2.4.20-28.9 and all is working fine. I didn't have time to work in
the issue but have seen it on two systems.
Anyone have any idea what the
Hello All!
I have successfully set up my Cisco 5350 for use with *!
Through direct-inward-dial i have all my users dialing my number placed
in Asterisk.
But I have a problem - one way sound (it IS NOT a codec issue):
When I call the 5350, it connects to the Asterisk, and then to the
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