Hi friends,
I used following commands to configure my zaptel card
1.modprobe zaptel
2.modprobe wct1xxp
3.ztcfg -vvv
4.zttool
the problem is when I type zttool command it shows
RED Digium Wildcard T100P T1/PRI Card 0
my zaptel.conf look like this
HI there !
Need help, I'm using asterisk 0.9.0,pwlib 1.5.2,openh323_1.12.2 and
asterisk_oh323_1.5. All H323 Endpoint can dial each other for 30 sec,
after that connection lost because of an H323 Control Protocol Error !!!
this is the asterisk output while phoneing :
On 13 Sep 2004, Murali wrote:
[snip]
the problem is when I type zttool command it shows
RED Digium Wildcard T100P T1/PRI Card 0
my zaptel.conf look like this
span=1,1,0,esf,b8zs
[snip]
can any one suggest how to get ok signal.
There are several possible error
Hallo Kevin
On Sun, 12 Sep 2004 22:25:41 -0400 you wrote:
I am using Galaxy Voice until recently I can receive any inbound calls.
Thats done via the register entry
If I remove the [galaxy voice] context in my sip file the call rings in
but I obviously can't make any outgoing calls. Any
Steve Maroney wrote:
Hey guys,
Im about to sign up for VoicePulse Connect. Of course, I plan on using my
asterisk server to register = with the service. I would rather sign up
with VoicePulse via SIP instead of VoicePulse Connect. My asterisk box is
behind another Linux box serving as my
Hi,
I am interesting how can I use the capabilities of ProSLIC to
measure the following PSTN parameters:
- Voltage (V) Polarity (+-);
- Current (A);
- Frequency (Hz).
Are there any ready for use tools? If there aren't ready for use
tools how can I do the above measuring?
On Wed, Aug 25, 2004 at 04:22:26PM -0700, [EMAIL PROTECTED] wrote:
I finally have my 7920 working though I'm seeing this bizarre
behavior. As soon as the 7920 boots and authenticates with the AP my 7960
release's its ip.
Hi,
I have exactly the same problem. Have you found a solution or
Hi to all,
i saw that in chan_sip there is the possibility to let the * to take the
number from the Remote-Party-ID header field on incoming calls from gateway.
What about to let the * to generate the Remote-Party-ID on outgoing calls?
this is is useful for us to let the users to have their
On Sun, 12 Sep 2004 17:50:14 -0300, Marcelo Pacheco [EMAIL PROTECTED] wrote:
Has anyone been able to sucessfully use 2-4 X101P clones with Asterisk on a
single system ?
I'm using one Encore MD 3200 modem sucessfully, alongside a TDM400P (3 FXS, 1
FXO) and would like to be able to use up to 4
On Sun, 12 Sep 2004 23:46:22 -0300, Thomas Hutton [EMAIL PROTECTED] wrote:
I'm curious where I can find a good document describing how to weave
together some servers in different places. Trying to keep things as
simple as possible here, I don't understand how to get 2 way calling
going on
hi all,
can anyone give solution for this.
wct1xxp - Digium Wildcard T100P T1/PRI Card 0
zttool gives
RED Digium Wildcard T100P T1/PRI Card 0
my zaptel.conf look like this
span=1,1,0,esf,b8zs
bchan=1-23
dchan=24
loadzone=us
defaultzone=us
the above 5 lines only
hi all,
can anyone give solution for this.
wct1xxp - Digium Wildcard T100P T1/PRI Card 0
zttool gives
RED Digium Wildcard T100P T1/PRI Card 0
my zaptel.conf look like this
span=1,1,0,esf,b8zs
bchan=1-23
dchan=24
loadzone=us
defaultzone=us
the above 5 lines only
Hi all,
I get Unknown RTP codec 72 received message in console when call in
progress, but all seems OK. Calls is from X-Lite SIP softphone to PSTN
over voicepulse connect (IAX) and to FWD echo test (SIP). But this
message only with one SIP client, others (X-Lite too) not giving this
message. All
Hi,
I've recently installed a new Asterisk server (CVS-09/10/04-20:02:38-CEST) and
I'm trying to connect it via IAX to an existing server. But I keep getting
the error below (ethereal). I've got 6 other working Asterisk servers
connected.
11:54:00.955083 a.a.a.a - b.b.b.b IAX2 IAX, source
On Fri, Sep 10, 2004 at 10:35:33PM +0200, [EMAIL PROTECTED] wrote:
On Sat, 11 Sep 2004 [EMAIL PROTECTED] wrote:
I posted to the -dev list the other night (although I was a little
drunk) about whether the busydetect code recognizes the cadences as
well as the tone. Reason being that there
On Fri, Sep 10, 2004 at 02:34:45PM +1200, Colin Haxton wrote:
Okay. I read that it should be wcfxs in a list somewhere, makes sense
that it's the card not the module though. :)
If I load it under a 2.4 kernel the pci comes out as
-
Bus 0, device 8, function 0:
I am interesting how can I use the capabilities of ProSLIC to
measure the following PSTN parameters:
- Voltage (V) Polarity (+-);
- Current (A);
- Frequency (Hz).
Are there any ready for use tools? If there aren't ready for use
tools how can I do the above measuring?
Hi,
After evaluate the different options,
Ihave decidedmake an attempt changing chan_sip.c to retrieve full
sip.conf from mysql database. Since Matthew may have made advances during last
weekend,It would be good have aquick report of your research.
Iwill also try to contact Ehud Gavron,
OK. Thanks a lot. I'll change my first set up then...:)
I want to be able to make a PSTN call to the line connected to asterisk,
and that asterisk answer that call and ask for a sip number to dial
is this also simple? can you give me a simple setup for this?
Thanks a lot for your help.
Eric
Good day all
I added the music on hold entry in vpb.conf and commented out default line in
musiconhold.conf.
Asterisk starts up with the default mp3 but as soon as I remove it and add my
mp3 it just doenst start up and gives a broken pipe error?
Please Help or advice
Thanks
ALtus
Hi again. I've used the zapata.conf and the extensions.conf files at the
adrees you sent (showed below). I didn;t use the sip.conf file because
it seems to be appliable only to sip devices... and here comes my first
doubt:
At the Getting Started link included at www.asterisk.org
Have you tried initiating a monitor command from within the AGI after the
DIAL action?
When you place a call from AGI to connect a SIP phone to a local extension
like that, it DOES treat it as a separate call until the Zap channel is
connected. You are basically recording the dialing and when
Make sure you don't use mp3 with variable bitrate.
Record mp3 at 128 - constant should work.
/ Stig Henning
- Original Message -
From: Altus Snyman [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]
Sent: Monday, September 13, 2004 1:53 PM
Hello!
Am Montag, 13. September 2004 13:53 schrieb Altus Snyman:
Good day all
I added the music on hold entry in vpb.conf and commented out default line
in musiconhold.conf.
Asterisk starts up with the default mp3 but as soon as I remove it and add
my mp3 it just doenst start up and gives a
In the howto it tells me I should strip the ID3 tags
How do I do that?
On Monday 13 September 2004 14:24, Andreas Roedl wrote:
Hello!
Am Montag, 13. September 2004 13:53 schrieb Altus Snyman:
Good day all
I added the music on hold entry in vpb.conf and commented out default
line in
Unfortunately, this doesn't really work out to be a great solution. The
dynamic range of the original recording is limited and scaling it after
the fact just yields fairly distorted sounding recordings.
It seems like the problem is a bug in the implementation of the
compressed file format
Hi Francesco,
you can easily run two Asterisk systems back to back in the way you
described below.
--- ---
| Asterisk 1 || Asterisk 2 |
|TE405 |==X=| TE405|
|Provide
Hi Everyone
I have done the following steps with asterisk are as follows:
1. I have installed latest version of asterisk from cvs.
2. I have used database as postgresql.
3. I have downloaded ast_data.tar.gz from
http://svn.asteriskdocs.org/res_data/. I have followed the steps as per the
Marcello,
This is something I am hoping for as well but I cannot find any features within the
code to allow Asterisk to modify/create this field. Ideally I'd like to see the
CallingPres function support Remote-Party-ID to disable/enable privacy. I actually
placed a feature request some months
http://developers.slashdot.org/developers/04/09/13/1058241.shtml
_
Surf the net and talk on the phone with Xtra JetStream @
http://xtra.co.nz/jetstream
___
Asterisk-Users mailing list
Hello!
Am Montag, 13. September 2004 14:40 schrieb Altus Snyman:
In the howto it tells me I should strip the ID3 tags
How do I do that?
http://fibiger.org/mp3tag.html
Andi
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
You don't have a T1 connected to the card or you
are using a wrong cable or... A red alarm says 'I don't like the signal
coming in from the far end'.
Could be no signal, could be incompatible signaling
options, could be wiring issues, could be most anything.
Lyle
- Original Message
I never stripped my tags and things work fine for me. I had problems at
first too with MOH. My problem was due to how I was copying over the
files. I was copy via FTP using the command line in Linux. However, if
you do not explicitly state binary as the copy method, it will copy the
files over
Followed; http://www.voip-info.org/wiki-Asterisk+Agents
agents.conf
[agents]
agent=1001,4321,BenDover
queues.conf
[queue1]
member=Agent/1001
extensions.conf
exten=28,1,AgentLogin(1001)
exten=29,1,Queue(queue1)
But when I call number 28, I get:
"Please enter your password followed by
Hi guys,
What is different and the context to
play unavail message and busy message?
When a SIP connection is unregistered, voicemail will
play unavail message, right?
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
On Sun, 2004-09-12 at 21:25, Kevin wrote:
I am using Galaxy Voice until recently I can receive any inbound calls.
If I remove the [galaxy voice] context in my sip file the call rings in
but I obviously can't make any outgoing calls. Any suggestions?
Don't remove the [galaxyvoice] entry from
On Mon, 2004-09-13 at 05:39, Murali wrote:
hi all,
can anyone give solution for this.
wct1xxp - Digium Wildcard T100P T1/PRI Card 0
zttool gives
RED Digium Wildcard T100P T1/PRI Card 0
my zaptel.conf look like this
span=1,1,0,esf,b8zs
bchan=1-23
On Mon, 2004-09-13 at 09:02, Jozeph Brasil wrote:
Hi guys,
What is different and the context to play unavail message and busy
message?
When a SIP connection is unregistered, voicemail will play unavail
message, right?
No. If there's a n+101 Asterisk will jump to that if the
On Mon, 2004-09-13 at 06:13, Elman Efendiyev wrote:
I get Unknown RTP codec 72 received message in console when call in
progress, but all seems OK. Calls is from X-Lite SIP softphone to PSTN
None of the 19 hits I saw on Google about this were helpful?
macroTo search the Asterisk mailing list
The subject says it all. A couple of my sons have very annoying friends
that tend to call ALOT. I usually don't like to answer the phone but
these kids keep calling back with in 2 minutes of calling. I'm sure
someone else has this problem and maybe using * to do a callerID match
and block?
Hi,
I have an IAXy which *appears* not to renew it's DHCP lease.
The DHCP server is a Solaris box running the native Solaris DHCP server.
Is there any known DHCP issues I should be aware of?
Is there anyway to get the IAXy to log it's DHCP activity?
Thanks,
Glenn
Ironic, Im just working on something similar myself, you can either use the
appropriately named ex-girlfriend feature or I use GotoIf statements to match the
caller id and maybe a timer or something to route to another context.
; note page search in girlfriend
Hi,
I have a RH 9 box just freshly installed on a Dell optiplex GX1. I also
have an quicknet phone jack isa card installed and is seen by the OS. I
checked out the latest cvs from the Asterisk site and compiled and installed
fine. I also downloaded the mpg123 tar.gz and compiled and installed
Hi!
I wonder what sondfile formats Playback() could play. I know it plays
GSM but to save the CPU time I will avoid converting GSM alaw for my
E1 and user alaw compressed wavs.
Unfortunately the wiki does not list the supported filetypes but I know
that there are a few.
Thanks in davance
___
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At the cli do
show file formats
bkw
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Christian Victor
Sent: Monday, September 13, 2004 9:42 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users]
___
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Hello list,
I'm developing a front-end for asterisk cdr-mysql and i need
a database for tests.
Could someone send me a mysql data for a test ?
I'm looking for a table with at least 5000 call records.
Best regards ,
-Jefferson Carvalho
___
According to IANA's list of RTP payload types
(http://www.iana.org/assignments/rtp-parameters) RTP payload type 72 fulls within the
following range:
72--76 reserved for RTCP conflict avoidance [RFC3550]
I can't find much else in RFC3550 that defines it further but this
Thanks for the hint Eric, but yes, before sending a message to the list
I checked google and wiki and NO - I didn't find an answer/solution/any
info on this subject.
There was couple of the same questions on the list but none of them
answered
--
Sincerely,
Elman Efendiyev
[EMAIL PROTECTED]
Brian West schrieb:
At the cli do
show file formats
Sorry - that does not work. And SHOW AUDIO CODECS shows me the codecs
but not the file formats (.wav etc) supported.
Christian
I wonder what sondfile formats Playback() could play. I know it plays
GSM but to save the CPU time I will avoid
On Mon, 2004-09-13 at 10:05, Christian Victor wrote:
Brian West schrieb:
At the cli do
show file formats
Sorry - that does not work. And SHOW AUDIO CODECS shows me the codecs
but not the file formats (.wav etc) supported.
Upgrade your Asterisk. show file formats works for Asterisk
I am still stuck, anyone got any ideas ?
Hi,
Having no digium hardware in my box and two cpus and a ohci usb bus im
forced to use zaprtc.
I have recompiled the kernel and removed enhanced rtc support.
When I attempt to compile zaprtc I get the following error.
zaprtc.c:1077: warning: implicit
Eric Wieling schrieb:
At the cli do
show file formats
Sorry - that does not work. And SHOW AUDIO CODECS shows me the codecs
but not the file formats (.wav etc) supported.
Upgrade your Asterisk. show file formats works for Asterisk
CVS-HEAD-07/18/04-11:25:14
I am running Asterisk
Update your asterisk install them because you must have an old one
asterisk*CLI show file formats
Format Name Extensions
SLINR slnsln|raw
ILBC iLBC ilbc
G726 g726-16g726-16
G726 g726-24g726-24
G726 g726-32g726-32
G726 g726-40
We're now opening up registrations for the Astricon tutorials again.
We've been able to move to new conference rooms within the same hotel.
Register on line at http://www.astricon.net
We're sorry for the inconvienience our recent closing of the tutorials
may have caused you. You are welcome to
Is this problem more-or-less continuous or does it happen occasionally?
If the latter, does it always happen on channel one?
At first glance, this looks like some sort of line hit, maybe a loss of
sync?
On Sep 10, 2004, at 8:58 AM, Claus Futtrup wrote:
Hi Guys,
Im having some problems with a
I've seen others complain about this error with no result. Any idea on this?
cvs server: failed to create lock directory for
`/usr/cvsroot/asterisk-addons/format_mp3'
(/usr/cvsroot/asterisk-addons/format_mp3/#cvs.lock): Permission denied
cvs server: failed to obtain dir lock in repository
On Mon, 2004-09-13 at 10:21, Brian West wrote:
Update your asterisk install them because you must have an old one
asterisk*CLI show file formats
Format Name Extensions
SLINR slnsln|raw
SLINR wavwav
SLINR mp3mp3
Is this correct or a bug in
On Mon, 2004-09-13 at 10:20, Christian Victor wrote:
I am running Asterisk CVS-09/12/04-16:39:35. I am quite new to CVS. Is
CVS_HEAD something different?
There was a difference at one time, not any more.
--
Eric Wieling * BTEL Consulting * 504-899-1387 x2111
In a related story, the
Does IAX2 properly update call records for transferred calls to another IAX2
server? Or should I still be using notransfer=yes ?
Example:
SERVER1 calls SERVER2 which transfers call to SERVER3
If Call records are pulled from Server2 will that call have proper CDRs?
The Wiki says no.
Low, Adam wrote:
Ironic, Im just working on something similar myself, you can either use the
appropriately named ex-girlfriend feature or I use GotoIf statements to match the
caller id and maybe a timer or something to route to another context.
; note page search in girlfriend
- Original Message -
snip
I've got 2x PRI 30 lines coming in to the Index, and I have 4 spare PRI
cards in the Index. I was thinking about using a QUAD PRI card from Digium
and having the calls come into the Index then transfer to Asterisk for IVR
then back to the Index. That way if
Its correct.. they all result in SLINR output.
bkw
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Eric Wieling
Sent: Monday, September 13, 2004 10:51 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE:
-Original Message-
From: Jason Kawakami [mailto:[EMAIL PROTECTED]
Sent: September 13, 2004 9:08 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Re: Astersk as AVAYA IVR
{clip}
Your assumptions on routing are correct. a path out of the
Index to the *
will require a path
I am trying to asses the possibility of setting up Asterisk at a collocation
service provider (ELI) for supplying VOIP to about 15 different
customers/locations. Is it possible to use one server to handle this load?
Is there a call management GUI (Windows), which will work well in this
situation
On Mon, 2004-09-13 at 12:08, C. David Kading wrote:
I am trying to asses the possibility of setting up Asterisk at a collocation
service provider (ELI) for supplying VOIP to about 15 different
customers/locations. Is it possible to use one server to handle this load?
It depends on how many
- Original Message -
snip
You may be thinking of PRI '2 B Channel Transfer' facility - see
http://www.voip-info.org/wiki-Asterisk+bounty+PRI+2B+channel+transfer
that is exactly what i was thinking. if it is on the bounty list i guess we
will wait for someone smarter than me to
On Mon, 13 Sep 2004 10:07:30 -0600, Jason Kawakami
will require a path back to the Index. not sure what you mean by
'tromboning' but it may be in reference to using the same b-channel on the
On tromboning and anti-tromboning, this is a feature that is part of
the Nortel ITG system we are
I am attempting to connect an asterisk system into an existing Network
Alchemy branch. This system supports h.323 and has an optional vocoder
card (Very expensive!) to enable other codecs.
I have achieved one-way audio in either direction but cannot get 2way. I
feel the problem is codec related.
Hello, everyone--
I've just posted a package on the Asterisk wiki that, when installed,
will allow you to announce incoming callers over the computer speakers,
based on their CID.
It's pretty simple, uses the linux /usr/bin/play (which on redhat, plays
gsm files just fine over the speakers), and
I Have addressed the scaling issue, according to one of the test cases I
read, a single server should be capable of handling 50 concurrent voice
calls. My main concern is its ability to run 15 different instances of * for
15 different customers simultaneously. Each client site should be able
Folks:
I'd like to install Asterisk for use in my home. However, I'd like to continue using
wireless phones in a couple of locations. The
cheapest way to do this is to continue to use analog phone devices via an FXO/FXS box.
However, I am not clear on whether I can
expect these devices to
[EMAIL PROTECTED] wrote:
I Have addressed the scaling issue, according to one of the test
cases I read, a single server should be capable of handling 50
concurrent voice calls. My main concern is its ability to run 15
different instances of * for 15 different customers simultaneously.
Each
[EMAIL PROTECTED] wrote:
Hello,
thanks for the answers!!! You mentionned to use the switch command. I
read about it in the WIKI, but I couldn't find enought information to
understand what it is actually doing. Can someone point me to the
right direction?
I would do this:
Create a single
US/Canada caller id can be forwarded between FXO to FXS devices.
On call waiting, I believe you need this: When I'm on my FXS and need to flash
the FXO device, I do flash then *0 on my phone. Works like a charm for me, I
use an asterisk behind another PBX, so when I need to transfer a call
On Mon, 2004-09-13 at 13:06, C. David Kading wrote:
I Have addressed the scaling issue, according to one of the test cases I
read, a single server should be capable of handling 50 concurrent voice
calls. My main concern is its ability to run 15 different instances of * for
15 different
I'm a bit new to the terminology. Let me ask my question more simply, even though I
think you already answered that it should
work
I want to receive calls into the Asterisk PBX via a cheap POTS-PBX method, such as a
WinModem or other FXS endpoint on the Asterisk
PBX.
I want the caller ID
Uniden's UIP300 transfer key is pretty much just a function key that can be assigned a DTMF value. How could I have asterisk monitor the channel for lets say *#*# then wait for a extension timeout then Start a consultive transfer? Through extensions.conf? Is this even possible?
Any help would be
We're hoping to have a digital rights managed ebook version of VoIP
Telephony with Asterisk by the end of 2004.
William
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of hank smith
Sent: Saturday, September 11, 2004 11:33 PM
To: Asterisk Users Mailing
The vocoder card has the ability to do both alaw and ulaw 64 codecs ala
asterisk
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Stuart
Mackintosh
Sent: 13 September 2004 18:27
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Alchemy branch integration,
On Mon, Sep 13, 2004 at 02:32:29PM -0400, Andrew said:
I'm a bit new to the terminology. Let me ask my question more simply, even though I
think you already answered that it should
work
I want to receive calls into the Asterisk PBX via a cheap POTS-PBX method, such as
a WinModem or
iH
have trouble getting the festival command to work. when i dial
extension i have set up i get
Sep 13 11:25:33 WARNING[344080]: app_festival.c:440 festival_exec: Festival
returned ER
festival does work correctly when i use it from the unix command line. my
festival.conf file is set up per
Quick question:
Are the hotel rooms registered automatically upon receipt of payment for
AstriCon plus hotel room costs ? Or do we have to register manually by
calling the hotel? Thanks -
On Monday 30 August 2004 10:07 pm, Steven Sokol wrote:
Just a brief reminder to everyone who wishes to
Hi all,
Is there a possibility to set the codecs Asterisk will choose in the dialplan
(exten= statements or their contexts) instead of sip.conf?
My problem is that I connect my SIP phone with several providers (Nikotel,
Sipgate, Stanaphone) for icoming and outgoing calls. Not all of these
Hi.
I have a x100p card installed and also asterisk, but I just dont get
asterisk to register with my sip provider (FWD)... when I start asterisk
using the following command I get the following messages (first, a lot
of messages show up immediatly after starting up: I'read this is normal,
then
Yep. I was on a 2.4 kernel to start with and that failed so I moved to
the 2.6 kernel hoping that would fix it.
Basically it fails in both. :(
Colin
Michael George wrote:
On Fri, Sep 10, 2004 at 02:34:45PM +1200, Colin Haxton wrote:
Okay. I read that it should be wcfxs in a list
You said what is possible and in exists.
Can you try with the settings what you have mentioned.
-Kannaiyan
- Original Message -
From: Andreas Greulich [EMAIL PROTECTED]
To: Asterisk-Users [EMAIL PROTECTED]
Cc: Greulich, Andreas [EMAIL PROTECTED]
Sent: Monday, September 13, 2004 8:35 PM
On Mon, 2004-09-13 at 15:31, Rich Allen wrote:
iH
have trouble getting the festival command to work. when i dial
extension i have set up i get
Sep 13 11:25:33 WARNING[344080]: app_festival.c:440 festival_exec: Festival
returned ER
festival does work correctly when i use it from the
Greetings,
A few days ago, I set up asterisk-1.0-RC2 on a Redhat 9 box.
I also signed up for broadvoice and am now trying to get asterisk to talk
to broadvoice and so on.
I've tried setting up my sip.conf in two ways:
--
register =
Uniden's UIP300 transfer key is pretty much just a function key that can be assigned a DTMF value. How could I have asterisk monitor the channel for lets say *#*# then wait for a extension timeout then Start a consultive transfer? Through extensions.conf? Is this even possible?
Any help would be
Hi!
I am trying to start Asterisk 1.0-RC1 with chan_capi.
Here the error:
---
Parsing '/etc/asterisk/modem.conf': Found
== Loading modem driver chan_modem_chan_capi.soSep 13 22:14:08
WARNING[1076968064]: loader.c:242 ast_load_resource:
/usr/lib/asterisk/modules/chan_modem_chan_capi.so:
I just bought an IAXy and have been using it with * and NuFone. Around
once every 24-48 hours of operation (not off hook, just powered up),
when I pick up the phone I hear loud static instead of dialtone. If
there's an incoming call, the phone will ring and * CLI will show that
it's trying
Hello Marconi,
still the same
got the following when setting up call
-- Executing Dial(SIP/home-422a, SIP/[EMAIL PROTECTED]) in new
stack Sep 13 22:52:48 WARNING[376851]: chan_sip.c:590 __sip_xmit:
sip_xmit of 0x815e6e4 (len 677) to 198.65.166.131 returned -1: Invalid
argument -- Called [EMAIL
I've tried setting up my sip.conf in two ways:
--
register = [240xxx]:[EMAIL PROTECTED]
[Broadvoice]
type=peer
username=[240xxx]
fromuser=[240xxx]
secret=[my_password]
host=sip.broadvoice.com
context=incoming
Hello Duncan,
Check your config against this one:
http://www.aspworld.com/telco/bv_sip.htm
--
Richard Cook
[EMAIL PROTECTED]
Tel: 705-497-9320 - ext 2010
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Monday,
On Mon, 13 Sep 2004 14:57:08 -0400, Walt Reed [EMAIL PROTECTED] wrote:
Check the asterisk Wiki for hardware compatability. I did not see an IP
- FXO (or FXS) device on the pcphoneline site. They had a couple IP
phones that may or may not work with * (check the wiki) and a USB FXS
adaptor that
Hi!
I am trying to start Asterisk 1.0-RC1 with chan_capi.
Here the error:
---
Parsing '/etc/asterisk/modem.conf': Found
== Loading modem driver chan_modem_chan_capi.soSep 13 22:14:08
WARNING[1076968064]: loader.c:242 ast_load_resource:
/usr/lib/asterisk/modules/chan_modem_chan_capi.so:
How do you configure an outbound proxy for Asterisk? If the
sip call is not local I want everything to go to a designated sip proxy.
Thanks,
Chad
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Asterisk-Users mailing list
[EMAIL PROTECTED]
I am running
Asterisk CVS-HEAD-08/25/04-20:11:31. I have two IAX accounts that I use,
FWD and VoicePulse. FWD requires you use ULAW codec to connect.
VoicePulse will work with most all the codecs Asterisk supports.
I recently tried to
force VoicePulse connections to use ILBC or GSM and
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