Alistair Cunningham wrote:
Michael Manousos wrote:
Alistair Cunningham wrote:
I have a customer who wants to do large volumes of H.323 to H.323
hairpinning. We haven't tested this scenario for large volumes
before; maybe someone on asterisk-users has.
If they buy a top of the line PC, how many
Hi there
I have a question regarding IAX jitter. I have 3 users on a
LAN dialing into a Meetme conference on an Asterisk box which is also hosted on
the LAN. I have set jitterbuffer = no and tos = lowdelay. Now, for 2 of the
users the audio is fine, but for the 3rd user there is
Hi,all
I am newer to Asterisk.My Asterisk version is the newest CVS-HEAD.now something appears in the console CLI like below these,I don't know what's happen to my Asterisk Server.Could anybody help me? Thanks
Junk at the beginningWarning, flexibel rate not heavily tested!Junk at the
Hi,
Try to cancel 'silence suppression' from
the 'other' source.
Asterisk does not support 'silence suppression'
(yet ?) (as far as I know)
Regards,
Shaoul Jacobson
Senior VoIP Consultant
Tellink
Tel : +32 3 201 96 36
Fax : +32 3 227 09 81
e-mail [EMAIL PROTECTED]
On 5/16/05, Steve Kennedy [EMAIL PROTECTED] wrote:
On Mon, May 16, 2005 at 03:51:28PM +0100, Mike Dent wrote:
Hi,
I'd be interested in comments from any users of the vonage service in the
UK?
http://www.vonage.co.uk is the website.
Where are the servers located, traceroute would be
Chris,
Don't forget that a change in features.conf requires a restart of
asterisk (or the modual features.c) - you can't get away with just a
reload.
On 5/17/05, Chris Mason [EMAIL PROTECTED] wrote:
Thanks, I removed that and will test. I don't have an analog extension here,
I am testing
lie ka skrev:
*Junk at the beginning
Warning, flexibel rate not heavily tested!
*
Do not use MP3's with VBR (variable bitrate).
Daniel
smime.p7s
Description: S/MIME Cryptographic Signature
___
Asterisk-Users mailing list
Hi list,
I have a problem with OH323. I follow the Joao's mail (thanks/obrigado
Joao!), but don't load chan_oh323.so.
When I run load chan_oh323.so, appear
May 17 10:44:33 WARNING[2513]: loader.c:258
ast_load_resource: /usr/lib/asterisk/modules/chan_oh323.so: undefined
symbol:
Romain Barrallon skrev:
- When I execute asterisk there is a warning and a noise in the headphones :
May 16 22:12:22 WARNING[3817]: chan_oss.c:269 sound_thread: Read error on sound
device: Resource temporarily unavailable
Does it come from the driver of the sound card ?
Do you really need sound
kurt x wrote:
Since the 3620 has two slots it should be configured as follows:
port slot #/0:D
This gives me:
delta(config-dial-peer)#port 0/0:D
^
% Invalid input detected at '^' marker.
delta(config-dial-peer)#
Kurt
Hello!
How can I check if oh323 is loaded and working? Is there a quick test for this?
Thank you.
Micko
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To UNSUBSCRIBE or update
On 16 May 2005, at 22:54, Jean-Denis Girard wrote:
Andres Paglayan a écrit :
File::copy does copy, it re-writes the file,
you need to move it.
so when the the pointer is placed the file is already there.
Well from File::Copy man page, about the move() function:
If possible, move() will simply
On 17 May 2005, at 02:21, Corey Hickey wrote:
Hello,
The company I work for deploys and manages telecom hardware for
small- to
medium-sized businesses. My boss has asked me to investigate
Asterisk as a
possible PBX for deploying to customers along with IP phones. The
general
layout would be:
I am planning to deploy an Asterisk server at a local restaurant and was
thinking: I hear a lot of troubles using fax machines with IP trunks.
What about using Credit Card readers? Same basic technology right? A
slow modem to negotiate the transaction. Does anyone have any caveats?
On Tue, 2005-05-17 at 09:26 +0100, tim panton wrote:
On 16 May 2005, at 22:54, Jean-Denis Girard wrote:
Andres Paglayan a écrit :
File::copy does copy, it re-writes the file,
you need to move it.
so when the the pointer is placed the file is already there.
Well from File::Copy
I think, that In C3600 platform, there is no 0:D port, but D channel is
named as 0:15.
So try port 0/0:15, if you want to use first E1 port in first slot of
router.
Anyway, try to use ? character. Look:
SIP-3640(config-dial-peer)#port ?
2-3 Voice interface slot #
Perhaps the params for the database connection arent correct!
Please check the host, username, password in the defines.php file.
Cheers
On Sun, 2005-05-15 at 09:49, [EMAIL PROTECTED] wrote:
Hi,
I have installed AreskiCC on Slackware 10.1 with Asterisk latest CVS
and Postgres 7.4.
On Tue, 17 May 2005, tim panton wrote:
The 'if possible' thing relates to filesystem design.
Almost all of the native UNIX filesystems support mv as an atomic action
- but only within the same filesystem.
(Imagine you create the file on one physical disk then 'move' it
onto a different disk
Hello,
Did anybody get working with rtptimout option set localy. (I tried with no
results).
I mean not in [general] but with specific peer definition like this:
[myfax]
type=friend
Secret=xxx
username=xxx
host=xxx
fromuser=xxx
insecure=very
mailbox=xxx
rtptimeout=9000
nat=no
language=pl
I had CC readers going over the internet (with pings over 80ms)
connected to Linksys PAP2.
It was only successful once every 3 attempts.
I had 100% reliability when it was connected on LAN.
Timing is an issue, if you doing everything on LAN it should not be a
problem. Just make sure you use G.711
Hi all
I am tying to restrict certain callers to call in with a designated PIN and then have asterisk record the PIN and the ANI number onto the CDRs. I created a .txt file with all the PINs and * let us call in and then dial the desired number without a problem. But refuse to insert the PINs in
Without on/offhook, I know there are USB 10-key pads out there, although
that isn't really what you're looking for I should think. The +, - or
ENTER key could be remapped to macro something else I should think...
Chris Coulthurst
[EMAIL PROTECTED]
|-Original Message-
|From: [EMAIL
The ^ marker appeared under the o in port, the email changed the
position :(
So it seems as if the port command in missing.
delta(config-dial-peer)#port ?
% Unrecognized command
delta(config-dial-peer)#port
Julian
barney wrote:
I think, that In C3600 platform, there is no 0:D port, but D
Michael Manousos wrote:
Alistair Cunningham wrote:
Michael Manousos wrote:
Alistair Cunningham wrote:
I have a customer who wants to do large volumes of H.323 to H.323
hairpinning. We haven't tested this scenario for large volumes
before; maybe someone on asterisk-users has.
If they buy a top
Here you can calculate it for yourself:
http://www.asteriskguru.com/bandwidth_calculator.php
Michael Welter wrote:
Would someone please show me how to calculate the required bandwidth
for 50 GSM channels on a _trunked_ IAX connection(s)? 100 channels?
What would be the packet size? Header is 12
GN-Netcom GN 7100
http://www.gn-netcom.com/US/EN/MainMenu/Products/Headset+Telephones/GN+7100.htm
Bill Hunt
Stroudwater Contact Point
www.stroudwater.com
Realize the Value of Customer Contact!TM
This e-mail is intended solely for the person or entity to which it is
addressed and may
Title: Background() problem (with queue(), etc.)
Hi all,
I want to be able to play messages while someone is in a queue. From the documentation the Background() command should work: From wiki: Starts playing a given sound file, but immediately returns, permitting the sound file to play in
On 17 May 2005, at 10:44, Peter Svensson wrote:
On Tue, 17 May 2005, tim panton wrote:
The 'if possible' thing relates to filesystem design.
Almost all of the native UNIX filesystems support mv as an atomic
action
- but only within the same filesystem.
(Imagine you create the file on one
Hi,
In order to save public IPs, I am attempting to use a Router SureConnect of US
Robotics in order to route calls to Asterisk on a private IP.
Would you recommand a large router like Cisco if we have 30 calls or a normal
router can do ?
Any advise is greatly appreciated
Hi all,
i've installed Asterisk with AMP. I've created 4 extensions (for 4 SIP phones)
and this is what my sip show users return:
moloch*CLI sip show users
Username Secret Accountcode Def.Context ACL NAT
204 moirafrom-internal
I have a question regarding IAX jitter. I have 3 users on a LAN dialing into
a Meetme
conference on an Asterisk box which is also hosted on
the LAN. I have set jitterbuffer = no and tos = lowdelay. Now, for 2 of the
users the audio is
fine, but for the 3rd user there is intermittent
although a great discussion, has anyone actually got the voicemail.conf
from a DB, the query that asterisk sends is
SELECT category, var_name, var_val, cat_metric FROM voicemail_users
WHERE filename='voicemail.conf' and commented=0 ORDER BY filename,
cat_metric desc, var_metric asc, category,
I'ld like to test SMSes between Asterisk and an analog phone on a
GS-ATA-286 (SIP). I have spent many hours trying any exemple on
voip-info or mailing-list, but no message got sent from the analog
phone. My goal, at now, is to send SMSes locally from and to Asterisk.
Does anyone have a working
I am planning to deploy an Asterisk server at a local restaurant and was
thinking: I hear a lot of troubles using fax machines with IP trunks.
What about using Credit Card readers? Same basic technology right? A
slow modem to negotiate the transaction. Does anyone have any caveats?
Wow :-). Ok, try to send whole running-config from your router and also send
me output from commands: show ver, show voice dsp and show inte desc
-b
- Original Message -
From: Asterisk [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Chris Mason (Lists)
Sent: Saturday, April 30, 2005 5:59 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] Problem with Sangoma/Adtran 600
I've been trying to set up incoming faxes using spandsp with a HFC card.
Unfortunately, incoming faxes are of very poor quality, the pages are
not transferred wholly (sometimes only a bit of a page is transferred etc.).
From what I've read, I may have troubles with correct timing (and
need to
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Paul wrote:
Anyway, bumping him is not extreme at all. IIRC - some lists are setup
to automatically unsubscribe people after N days of delivery failures.
We only see this individually when we post but the list server is
probably getting this for
Doubtfull the 99.98% is any sort of problem. The number would have to be
much lower to have a constant impact on audio.
A kernel recompile managed to give me an average of 99.98%, but I am not
certain that this is optimal. It seems that I get some background hiss,
Yet another alternative would be to process credit cards over the
Internet. Of course then you need a PC at the POS, but Internet card
processing is much faster than using the modem in a stand alone
terminal (no dial/negotiation for each transaction)
BTW, from my experience with fax, I would not
I am planning to deploy an Asterisk server at a local
restaurant and was
thinking: I hear a lot of troubles using fax machines with IP trunks.
What about using Credit Card readers? Same basic technology
right? A slow modem to negotiate the transaction. Does
anyone have any caveats?
I
Thanks for the advice. That was part of the problem, but also the Sangoma
stuff required I removed the modules from /etc/modules.conf and I had to
configure the DSO mapping in the channel bank, which I knew nothing about.
The Sangoma T1 card and the Adtran Channel Bank now work very well together,
Many thanks for your help: Details are as follows:
Show voice dsp =
delta#show voice dsp
DSP DSP DSPWARE CURR BOOT PAK
TX/RX
TYPE NUM CH CODECVERSION STATE STATE RST AI VOICEPORT TS ABORT
PACK COUNT
=== ==
It is indeed much higher. I'm using it here in production w/o media
running through it and it is supporting 400 connections with virtually
no load on it on a 1.8Ghz machine.
On 5/17/05, Alistair Cunningham [EMAIL PROTECTED] wrote:
Michael Manousos wrote:
Alistair Cunningham wrote:
Michael
I can`t see voice-port in your configuration. Something like this:
!
voice-port 1/0:15; voice-port 1/0:15, if your D channel is
Serial1/0:15
input gain -6
output attenuation 14
echo-cancel coverage 32
echo-cancel suppressor
cptone SK
description E1
bearer-cap Speech
!
If you
Hi Everyone!
Is there any hope for us newbie plebs who
want to also get hold of the updated Cisco firmware?
I need to get a 7910G updated to work on
SIP..
Any help on obtaining the updated firmware
quickly and painlessly would be great J
Cheers
M
From:
[EMAIL PROTECTED]
Hello,
I need to implement Asterisk and Digium cards for
intra-office and international calls.
PSTN ---SS7--- Asterisk+Digium ---IP--- SER ---IP---
Asterisk+Digium ---SS7--- PSTN
Can I please have an answer about the following questions?
How many Wildcard TE410P or
TE405P
Peter Svensson wrote:
On Mon, 16 May 2005, Steve Underwood wrote:
It is possible, though complicated, to synchronize the 2Mbit clocks on two
unrelated cards by measuring the accumulated phase shift (difference in
interrupt rate) over time and compensating, thus implementing a PLL in
It doesn't help at all, since you are talking rubbish. Try to keep track
of the subject matter. We are discussing modems, where not slipping is
vital.
Regards,
Steve
Rich Adamson wrote:
It doesn't make any difference. The pcm data that arrives from the telco
is buffered in the zaptel and/or
Hi everybody,
Recently my Asterisk server started to behave strangely:
in some cases (hard to diagnose and reproduce), the SIP module stops
responding, and the log is filled with messages Failed to grab lock,
trying again... (about a hundred messages or so per second).
This often
Title: Background AGI
Hello. I'm wondering if it's possible to have an AGI script run in the background of a normal call (not a MeetMe room).
What I'm ultimately trying to achieve is that an ACD agent on a SIP extension (or any other technology for that matter) is able to use DTMF codes to
Mark Brown wrote:
Hi Everyone!
Is there any hope for us newbie plebs who want to also get hold of the
updated Cisco firmware?
I need to get a 7910G updated to work on SIP..
Any help on obtaining the updated firmware quickly and painlessly
would be great J
Cheers
M
7910 does not have a SIP
CLI show module
and look for chan_oh323.so
If oh323 is loaded, oh323 show conf will provide more useful info.
On 5/17/05, Micko [EMAIL PROTECTED] wrote:
Hello!
How can I check if oh323 is loaded and working? Is there a quick test for
this?
Thank you.
Micko
Hi Jean-Yves,
As a new topic it would have been nicer to everyone to start a new thread.
Are you talking about sending or receiving faxes?
txfax can insert a header on each page when it sends a fax. There is a
parameter for that.
When a fax is received the header you see is part of the image. As
Hello.
, , ,
,
+ .
I wrote a small perl script, that just calls to the specified number and then
receives the information about
+the status of the call.
This script is below:
#!/usr/bin/perl
use Asterisk::AGI;
$AGI = new Asterisk::AGI;
my %input = $AGI-ReadParse();
Hi Everyone!,
Im not sure if my previous request went through or
notIt got returned as not-deliverable??? Who knows!
Anyway, I need to get hold of the Cisco Firmware to upgrade
a 7910G to sip.
I know it can be a real pain in the butt getting hold of the
firmware, so any help in
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Sorry, the 7910(G) only supports SCCP/skinny ate present
On Tue, 17 May 2005, Mark Brown wrote:
Hi Everyone!
Is there any hope for us newbie plebs who want to also get hold of the updated
Cisco firmware?
I need to get a 7910G updated to work on
Hi everybody,
Recently my Asterisk server started to behave strangely: in some cases (hard
to diagnose and reproduce), the SIP module stops responding, and the log is
filled with messages Failed to grab lock, trying again... (about a hundred
messages or so per second).
This often happens at the
Steve, the stuff below was in direct response to a user question
regarding T1 timing sync and understanding why that might be
important relative to spandsp and/or other modem use. So it does
apply directly. (Of coarse it slid into a tangent.)
It doesn't help at all,
Mark,
As far as I know, the 7910 does not have SIP firmware.
Doug
Mark Brown wrote:
Hi Everyone!
Is there any hope for us newbie plebs who want to also get hold of the
updated Cisco firmware?
I need to get a 7910G updated to work on SIP..
Any help on obtaining the updated firmware quickly and
Asterisk (or at least Asterisk - Stable) will not present DNIS to the CLI.
If the number originally dialled is redirected then you will see the
redirected number presented rather than the original one. The original
number appears to be present in libpri but doesn't seem to make it into
asterisk
ok, I'm starting to get confused, you must be getting annoyed .. how do
I add this voice-port ?
Also, if I want to use * to handle all of the SIP internal calls, and
cisco to handle to inbound and outbound isdn-32 PRI calls, what feature
set of the IOS should I need ? What hardware on the
Thanks for that Mark... doesn't sound promising then :(
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mark
Johnson
Sent: 17 May 2005 14:01
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Cisco contract for
Tomasz Chmielewski wrote:
I've been trying to set up incoming faxes using spandsp with a HFC card.
Unfortunately, incoming faxes are of very poor quality, the pages are
not transferred wholly (sometimes only a bit of a page is transferred
etc.).
I tried sending faxes from different fax devices,
On Tue, 17 May 2005, Steve Underwood wrote:
In most hardware the clock you use is not provided by a crystal. Rather
the crystal provides a reference for a pll. The conversion factor between
the crystan and the derived clock is usually tunable.
Nope. Its always a crystal. Its either a
Hi there
I have users that are using IAX clients, dialling into
meetme conferences. They will be on varying connection speeds. Firstly, should
jitterbuffer be used with meetme? Secondly, I have read some posts which
indicate that jitterbuffer is not that stable. Is it stable enough to
Hello,
I need to implement Asterisk and Digium cards for
intra-office and international calls.
PSTN ---SS7--- Asterisk+Digium ---IP--- SER
---IP--- Asterisk+Digium ---SS7--- PSTN
Can I please have an answer about the following questions?
How many Wildcard TE410P or
TE405P
Hello everybody,
Does anybody know how do we
configure the asterisk realtime extensions using Mysql or ODBC. I have gone
through the information given on the site www.voi-info.org,
but didnt get thru.
Would you please tell me as to how
do I go about configuring the asterisk realtime
channels? I've got a TDM422P and since I know the phone numbers
associated with each of the 2 FXO channels I'd like to set that
so that future extensions contexts can use it and the caller-id
info in the form mydidnum/callerid like I can with VOIP DIDs.
I haven't found if there is a variable
You need an NM-HDV card of some sort to run voice. The WIC-1MFT-E1 can
handle voice, but you still need the DSP's to use it as a voice card.
Putting that into an NM-HDV that has DSP's will make the voice ports and
dsp's show up.
Asterisk wrote:
ok, I'm starting to get confused, you must be
How can I increase this timeout?
CLI show application dial
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Hi Sharath,
For configuring the sip mysql peers go thru the details given in
the site www.voip-info.org for sip mysql peers. Create the db table as given
on the site. And don't forget to set the autocreatepeer variable to yes in
the general section of the sip.conf file. Alright... this
Forget the dialogic, the drivers are old and not free and almost no-
one is using them.
I concur that view. Don't even touch Dialogic garbage. I have see people
spending months without being able to make them work. Dump them into
your incinerator and turn the knob from 'Medium' to 'High'.
Hi, We have a scenario where we receive calls from 2 different places:
1- Avaya IP Office
2- CIC Interactive Intelligence PBX
and the calls are transfer automatically to an Asterisk Box.
The problem we are experiencing is that more that half of those calls
come with Echo and Jitter. For outbounds
How can I setup my server to peer with other friendly networks.
I saw on FWD, that with a prefix of **NNN you can go to another
(friendly) network. Appending the users phone number would reach this
number.
How can I set this up with some friendly networks?
In the case of FWD, I could use my own
We may have gotten ahead of ourselves - if we have a PRI line and a linux
server running asteriks, is that the only hardware we need to forward out to
cell phones and to figure out which phone picks up a given call?
Thanks,
Theo
- Original Message -
From: Paul [EMAIL PROTECTED]
To:
Hi All,
Im having some trouble getting Asterisk to send DTMF via rfc2833. The
scenario is this:
For purposes of testing software, I have two applications communicating
with each other via DTMF. In between the two applications sits an
Asterisk. The applications require that DTMF be sent via
I don't have any problems using a pots line with the credit cards. In fact I
have in some locations a Sipura that is attached to the cc machine. Works
just set it up just like a fax using ulaw only.
Ariel
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
I have a problem when dialing from outside line to sip server. I get this
output on debug.
Could someone give me a hint what could be wrong?
== Starting OH323/R30149 at from-pstn,6000622,1 failed so falling back to
exten 's'
== Starting OH323/R30149 at from-pstn,s,1 still failed so falling
SS7 is a *signaling* standard, much like SIP, MGCP, H323. It is not
an encoding standard like PCM,g.711, g.729. Asterisk does not
currently support SS7. You could buy a SS7 - SIP (SIP-7) service
from a couple different service providers. Then you could use
Asterisk to take in your IMT
Skype is very succesfsfull and get more and more users, ... we can
ignore them, accept them or do something,...
My suggestion is that we try to do something, ...
If we would peer to each other, than we get soon also a great amount of
users together, and than our service becomes more valuable,
I have a problem when dialing from outside line to sip server. I get this
output on debug.
Could someone give me a hint what could be wrong?
== Starting OH323/R30149 at from-pstn,6000622,1 failed so falling back to
exten 's'
== Starting OH323/R30149 at from-pstn,s,1 still failed so falling
On Tue, 2005-05-17 at 20:02 +1000, Boris Bakchiev wrote:
I had CC readers going over the internet (with pings over 80ms)
connected to Linksys PAP2.
It was only successful once every 3 attempts.
I had 100% reliability when it was connected on LAN.
Timing is an issue, if you doing everything
On Tue, 2005-05-17 at 11:35 +0100, Seb Auriol wrote:
Hi all,
I want to be able to play messages while someone is in a queue. From
the documentation the Background() command should work: From wiki:
Starts playing a given sound file, but immediately returns,
permitting the sound file to
Good day all
I installed Fedora core3
I also installed mpg123 0.59r
but asterisk does not want to play anything..on 2 of my server
No BAckgroung,Voicemail..nothing
Never had this before
In the cli it shows its playing it
But nothing happens?
Please Help
Hello.
I was just brainstorming for a future project and was hoping to get some
creative ideas from the list. If I have multiple * servers at multiple
locations all connected together with a nicely partitioned dialplan (2XX for
office 1, 3XX for office 2, etc.) it's pretty straightforward to link
On Tue, 2005-05-17 at 14:30 +0100, Mark Brown wrote:
Thanks for that Mark... doesn't sound promising then :(
7910 does not have a SIP image and looks like it never will. I have
about 40 of these stupid things that I can't get to work 100% with
skinny or sccp. If you ever figure out how, be
On 5/17/05, Brian Roy [EMAIL PROTECTED] wrote:
On 5/16/05, Nathan Pralle [EMAIL PROTECTED] wrote:
Hi all.
I'm curious to hear about other people's HOME usage of Asterisk. Do you
have a really neat setup for home use? Fun stuff with VM and/or
forwarding and custom scripts?
If you
On Tue, 17 May 2005, Jean-Yves Avenard wrote:
Side questions about spandsp... Is it possible to print the fax
header like what most faxes do (that is: who is sending the fax, how
many pages are included etc...) I'm not talking about printing
callerid, often I receive fax from the US
Hi, Ive been using asterisk tapi from omni for a few
days now with outlook and it rocks, thanks guys (you should set up a paypal
account, even if you dont want the cash nominate a charity or
something).
Anyway long story short, Im using [EMAIL PROTECTED] and I
would like from time to
Ronald Wiplinger wrote:
Skype is very succesfsfull and get more and more users, ... we can
ignore them, accept them or do something,...
My suggestion is that we try to do something, ...
If we would peer to each other, than we get soon also a great amount
of users together, and than our service
One thought (other than having multiple routes, we're assuming there
just isn't one to '305' here) could be to allow
intersystem-wide-and-KNOWN extensions to still receive their voicemail
by caching it locally with a default Allison voice for later delivery
attempts, not unlike email servers can
Hi,
I have read a lot about the thread of faxing support in Asterisk as
well as spanDSP. However, either I don't fully understand other
people's applications or may be what I'm trying to do is different
from what others are trying to do.
I have a very simple setup. I have an asterisk server
Hi Gurus.
I searched the lists, wiki and the rest of the web but I still do not
understand this.
My Setup is as follows:
[ISDN via chan_capi or IAX2 DiD Provider] = [* PBX] = [IAX2 Clients
(Atcom AT-320ED)]
I want to get callgroup/pickupgroup and callwaiting working on the IAX
phones. Some
How can I set this up with some friendly networks?
On your side, assuming your extensions are numbers 2XX.
[outgoing]
exten = *393.,1,Dial(SIP/fwd.pulver.com/${EXTEN:4})
[default]
exten = _2XX,1,Goto(internal,${EXTEN},1)
On their side, ask them to send calls using your prefix, for example
On Tue, May 17, 2005 at 10:45:52PM +0800, Ronald Wiplinger wrote:
Skype is very succesfsfull and get more and more users, ... we can
ignore them, accept them or do something,...
My suggestion is that we try to do something, ...
If we would peer to each other, than we get soon also a great
I have a PRI from Bellsouth going to my asterisk box with a Digium
Wildcard TE110P. I would like to be able to use call forwarding without
having to use two channels. Is it possible to use call redirect with a
PRI. Does the BRIstuff package help at all?
Thanks,
Lenny Sawyer
--
Lenwood
Title: Message
Hi
all,
Of course I am a newbie, so please bear with
me...
I'm having a lot of trouble getting things to work
properly and I am sure it is a configuration issue somewhere, I'm just not sure
where...I have been all through my extensions.conf and cannot seem to see a
I'm pretty new to astersik and I need to know where I
could find a web page/doc where is described the usage
of ODBC within asterisk and what should be installed.
Yes I visited wiki, but it seams to be cdr centric.
I need to use asterisk in a context other than cdr,
such as an ivr with bus
Hi,
Any recommendations for PRI providers in San Francisco?
I'm not looking for the cheapest provider, but a reliable provider with
reasonable prices.
Thanks,
Justin
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