Hi all!
How do I make Asterisk recognize fax calls and disconnect them?
Regards,
Evert
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Maxim Vexler wrote:
On 7/6/06, Evert Meulie [EMAIL PROTECTED] wrote:
Hi all!
How do I make Asterisk recognize fax calls and disconnect them?
Regards,
Evert
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*bump*
Anyone? I still can't find little/no info on DNID... :-/
Regards,
Evert
Evert Meulie wrote:
Hi all!
I'm in the process of configuring an Asterisk server here that, based on
which number was called, should send calls to different extensions:
913 - 1 - ext. 1
913 - 2
Hi all!
I'm in the process of configuring an Asterisk server here that, based on which
number was called, should send calls to different extensions:
913 - 1 - ext. 1
913 - 2 - ext. 2
913-1 913-2 being 2 (of the) numbers we have coming in to our system
via our VoIP hosting
Hi all!
I'm in the process of configuring an Asterisk server here that, based on which
number was called, should send calls to different extensions:
913 - 1 - ext. 1
913 - 2 - ext. 2
913-1 913-2 being 2 (of the) numbers we have coming in to our system
via our VoIP hosting
Have you checked
http://www.voip-info.org/wiki/view/Asterisk+High+Availability+Solutions
Regards,
Evert
[EMAIL PROTECTED] wrote:
Hi all,
I would like to know if there is a solution to this question.
Scenario:
Two asterisk servers connected across the Intenet ( in SIP or IAX mode, no
As soon as they port it to Gentoo I'll try it out... ;-)
Evert
Kerry Garrison wrote:
Everyone should simply uninstall Skype and switch to the Gizmo project
because it interfaces quite nicely with Asterisk.
Kerry Garrison
Publisher - http://GeekGazette.com - http://VOIPSpeak.net
(949)
Just wondering...
Has someone ever contacted Skype/Ebay and asked them about their point of
view/opinion on interfacing with SIP / Asterisk? 8-)
Regards,
Evert
[EMAIL PROTECTED] wrote:
I sincerely believe that it's completely non-sense to make a channel for
Skype.
Skype is a
I found the price. $450 :-/
Kevin P. Fleming wrote:
Evert Meulie wrote:
That unit looks VERY promising! Thanks! :-)
Would anyone happen to know an approx. price for a unit like this?
Anyone? I bet the manufacturer of the unit would know a price for it,
and it's probably even
Hi all!
I am aware of products like http://www.rsdevs.com/psgw_sip.shtml which act as a gateway, but what I'd really like is a for example an Asterisk module that can route calls to Skype, perhaps the same
principle as IAX2?
I'm assuming more people are interested in this, but... does it
Hi all!
I am looking for a device that I can stick in a USB-port on my Asterisk server and that allows me to connect one/more (cordless) PSTN-phones in such a way that they'll work with SIP/Asterisk. I know
there are USB-phones, but what I'm looking for is 'the USB-phone without the phone', if
That unit looks VERY promising! Thanks! :-)
Would anyone happen to know an approx. price for a unit like this?
Regards,
Evert
BJ Weschke wrote:
On 12/16/05, Evert Meulie [EMAIL PROTECTED] wrote:
Hi all!
I am looking for a device that I can stick in a USB-port on my Asterisk server
Found it! For some reason [EMAIL PROTECTED] had chosen to build itself without
app_meetme.so!
After building this module by hand, all worked! :-)
Evert
Evert Meulie wrote:
Read before you reply... ;-)
To be 100% clear on zaptel/ztdummy, here's the output of my lsmod:
[EMAIL PROTECTED
Hi all!
I have an [EMAIL PROTECTED] 2.1 setup here which is working 99% the way it
should. The only thing that does not work is Meetme/Conferences...
In the log-file I see:
Dec 8 11:51:28 WARNING[3288] pbx.c: No application 'MeetMe' for extension
(from-internal, 8125, 6)
This is when I
/ztdummy modules installed ?
Kunal
On 12/8/05, *Evert Meulie* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
Hi all!
I have an [EMAIL PROTECTED] 2.1 setup here which is working 99% the way
it should. The only thing that does not work is Meetme/Conferences...
In the log-file
Turns out my VoIP provider made a booh-booh... ;-)
Evert Meulie wrote:
Hi all!
Quite a mystery. The following happened when I was on holiday, and no one
else has changed any configs of either Asterisk or the Cisco's in the
building...
The situation: Incoming works fine on all
Hi all!
Quite a mystery. The following happened when I was on holiday, and no one else
has changed any configs of either Asterisk or the Cisco's in the building...
The situation: Incoming works fine on all phones. Outgoing only works from
non-Cisco phones. When calling from a Cisco phone to an
Hi all!
Who can tell me what the correct/preferred/only DTMFmode setting is for
Windows Messenger SIP clients?
Regards,
Evert
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Hi all!
We use a PHP-portal for management of our projects contacts. Now I
would like to make it possible to dial contacts directly from the portal.
Since users have to log in, I can use that to determine which office
phone the call should originate from. And the number-to-be-dialed is of
to your
telephone system, set up the click to dial, and integrate it with your
PHP - email me off list.
Alistair Cunningham,
Integrics Ltd,
Telephony, Database, Unix consulting worldwide
+44 (0)7870 699 479
http://integrics.com/
Evert Meulie wrote:
Hi all!
We use a PHP-portal for management of our
Hi everyone!
I wonder whether the following would be possible:
Can Asterisk show the country from which a call originates on the
display, along with the phone number?
Regards,
Evert Meulie
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Asterisk-Users
Is there a page/site where the progress/info on this project is to be
found? :-)
Regards,
Evert Meulie
Jon Radon wrote:
Right now, you'd need an FXS port and a modem for HylaFax to use.
It's not an ideal setup, but more reliable than using an ATA such as
the Sipura. Steve Underwood
508) to 0.5.0.4 returned -1: Invalid argument
Who can tell me what causes these, and how to fix it...?
Regards,
Evert Meulie
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ping 0.5.0.4
connect: Invalid argument
Nope! ;-)
Andreas Sikkema wrote:
[EMAIL PROTECTED] wrote:
WARNING[245775]: chan_sip.c:597 __sip_xmit: sip_xmit of 0x3ddfa3a8
(len 508) to 0.5.0.4 returned -1: Invalid argument
Who can tell me what causes these, and how to fix it...?
Is that a valid IP
Hi everyone!
I have an Asterisk server here that also has Hylafax installed on it.
What else do I need to have that server send/receive faxes?
Regards,
Evert Meulie
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Hi!
When I call a colleague of mine from my Cisco (via Asterisk), they get
on their display:
From Evert
asterisk
How do I remove/change the 'asterisk' part?
Regards,
Evert
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[EMAIL PROTECTED]
Found it. It's a Micronet-specific error message. So much for
standards... :-/
Evert Meulie wrote:
From a 'sip debug':
Sip read:
SIP/2.0 100 Trying
From: Evertsip:[EMAIL PROTECTED] ext. IP];tag=as6e18534e
To: sip:[dialled [EMAIL PROTECTED] server of VoIP provider]
Call-ID: [EMAIL PROTECTED
Hi everyone!
I'd like to create the following: a user picks up the phone (gets a dial
tone), dials '0' for an 'outside' line, gets a second (different?)
dialtone, and is able to enter an external phone number.
How do I implement this in extensions.conf...?
Regards,
Evert
Maurizio Marini wrote:
On Friday 17 September 2004 11:43, Evert Meulie wrote:
How do I implement this in extensions.conf...?
maybe this may help...
http://lists.digium.com/pipermail/asterisk-users/2004-February/036737.html
Thanks! That works like a charm! The only thing I'd like to do now
Hi everyone!
The following: Any calls coming in on extension 12121212 should get a
message telling them to dial the extension of the person they are trying
to reach, and then press #.
The call should then go to the entered extension.
This is as far as I got...
Hi everyone!
Situation: when I call from cell phone to a asterisk-connected phone,
all works fine. When I call from the asterisk-connected phone (a Cisco
7960 SIP) to the cell, the connection gets made, but there is no audio
going in either way...
Asterisk reports the following:
Sep 16 08:27:41
Hi everyone!
Is it safe to use this (old!) webmin module with asterisk 1.0rc2?
Regards,
Evert
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Hi everyone!
I was wondering... Does the musiconhold quality improve if the mpg123
processes run with a negative priority? If so, is there a way to make
them start like that, so I don't have to renice them?
Regards,
Evert
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Dave Cotton wrote:
On Thu, 2004-09-16 at 09:35 +, Murali wrote:
Hi friends,
Redhat9.0 installed in my machine.Asterisk1.0 installed in that machine.
Thereis no mpg123 player. So, I download the mpg123 player and installed it.
My sound card is configured correctly.
When I tried to
Dave Cotton wrote:
On Thu, 2004-09-16 at 12:00 +0200, Evert Meulie wrote:
Dave Cotton wrote:
On Thu, 2004-09-16 at 09:35 +, Murali wrote:
Hi friends,
Redhat9.0 installed in my machine.Asterisk1.0 installed in that machine.
Thereis no mpg123 player. So, I download the mpg123
Thanks, that did the trick! :-)
Kinda weird though that the mp3's that actually come with Asterisk don't
work correctly 'out of the box'. Or is this a mpg123 bug?
Regards,
Evert Meulie
Andreas Roedl wrote:
Hello!
Am Dienstag, 14. September 2004 19:21 schrieb Evert Meulie:
Found new ID3
Hi everyone!
I now have obtained a couple of SIP-accounts at a local VOIP-provider.
How do I specify that ALL outgoing calls to _NXXX go out via one of
these accounts?
Regards,
Evert
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[EMAIL PROTECTED]
Hi all!
I'm trying to have asterisk route all outgoing calls out via my VOIP
provider.
exten = _NXXX,1,Dial,SIP/[EMAIL PROTECTED] seems to have them to in
the direct direction. However, debug shows that my asterisk doesn't
identify itself correctly:
Sip read:
SIP/2.0 100 Trying
From:
From a 'sip debug':
Sip read:
SIP/2.0 100 Trying
From: Evertsip:[EMAIL PROTECTED] ext. IP];tag=as6e18534e
To: sip:[dialled [EMAIL PROTECTED] server of VoIP provider]
Call-ID: [EMAIL PROTECTED] ext. IP]
CSeq: 102 INVITE
Via:SIP/2.0/UDP [my ext. IP]:5060;branch=z9hG4bK4fd1045b
Content-Length:0
7
prviders need this parameter
In your extension.conf add the following entry:
exten = _NXXX,1,Dial(SIP/[EMAIL PROTECTED],30,r)
This config is only for outgoing calls.
On Tue, 14 Sep 2004 10:01:37 +0200, Evert Meulie [EMAIL PROTECTED] wrote:
Hi everyone!
I now have obtained a couple of SIP-accounts
When I do a 'asterisk -vc' I get following, but asterisk does NOT stay up:
== Parsing '/etc/asterisk/asterisk.conf': Found
== Parsing '/etc/asterisk/extconfig.conf': Found
between both systems.
Should bindaddr (iax.conf) or externip (sip.conf) be defined for a setup
like this one?
Regards,
Evert
Do you know where it got the 10.138.3.2 IP from? Is it configured
anywhere on the server? Do you have
externip defined in that config file?
Evert Meulie wrote:
Hi everyone
Hi everyone!
Is there a way to let Asterisk connect to/interface with the Micronet
SP5210 SIP server ( http://www.micronet.info/Products/voip/SP5210.asp )?
It does not support IAX, but maybe there is another way...?
Greetings,
Evert Meulie
traffic going over these lines has no problems with this. The
192.168.2.x 192.168.11.x networks are fully 'connected' to each other...
Who knows the answer...?
Regards,
Evert Meulie
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[EMAIL PROTECTED]
http
Hi everyone!
I have a problem... We have received a couple of phone numbers for voip
from a local voip-provider. The work fine directly with a Cisco 7960,
but so far I've not been able yet to integrate them into Asterisk.
I've tried:
/etc/asterisk/extensions.conf
*
[ip-incoming]
then for
routing incoming, outgoing or both via this voip-provider?
Greetings,
Evert
Benjamin on Asterisk Mailing Lists wrote:
On Wed, 08 Sep 2004 10:08:00 +0200, Evert Meulie [EMAIL PROTECTED] wrote:
I have a problem... We have received a couple of phone numbers for voip
from a local voip
Hi!
It turns out my provider uses the Micronet SIP server. Any possibilies
to let this one interface with Asterisk?
Regards,
Evert
Evert Meulie wrote:
Hi everyone!
I have a problem... We have received a couple of phone numbers for
voip from a local voip-provider. The work fine directly
Hi!
If I install a CAPI-compatible ISDN-card in my server, will that:
a) enable me to connect that server to the public phone net
b) allow me to connect an ISDN phone to the server and use it as a SIP-phone
c) all of the above?
Regards,
Evert
]
[mailto:[EMAIL PROTECTED] On Behalf Of Leif Madsen
Sent: 28 July 2004 16:50
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Access voicemail from Cisco 7960
On Wed, 28 Jul 2004 14:22:17 +0200, Evert Meulie [EMAIL PROTECTED] wrote:
Hi everyone!
Who can tell me how I can access my voicemail? When I
PROTECTED]
Subject: Re: [Asterisk-Users] Outgoing works, incoming doesn't...
Evert Meulie wrote:
Hi!
Problem with my 7960. Outgoing calls work, but incoming don't. A 'sip
show peers' gives:
Name/usernameHostDyn Nat ACL Mask Port
Status
105/105
Hi!
Problem with my 7960. Outgoing calls work, but incoming don't. A 'sip show
peers' gives:
Name/usernameHostDyn Nat ACL Mask Port
Status
105/105 192.168.2.175D 255.255.255.255 5060
UNREACHABLE
Is there something wrong with the config on that
Hi everyone!
Who can tell me how I can access my voicemail? When I dial the voicemail on
my Cisco 7960 I get access, but when trying to enter my mailbox number it
seems that Asterisk doesn't 'get' any of the keys I press. DTMF problem
perhaps?
Any suggestions on how/where to fix this...?
')
Regards,
Evert
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Olle E.
Johansson
Sent: 28 July 2004 15:18
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Outgoing works, incoming doesn't...
Evert Meulie wrote:
Hi!
Problem with my
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