Hi list,
I'm trying to get some ideas on this subject.
Normally astersik sends emails with voicemail attached trough local MTA.
As far as i know there is no way for asterisk to authenticate to an external
mailserver to relay these emails.
Well, these days every provider has some sort of spam
Am Mittwoch, den 05.09.2007, 22:58 -0700 schrieb satish patel:
Dear all
I have FAX machine connected with audiocode SIP device
i am trying to send fax and when negosiation going on and i start send
fax button then my after half page it got stuck in fax machine so is
there any
Am Donnerstag, den 06.09.2007, 10:16 +0530 schrieb Benjamin Jacob:
Hello ppl,
Any way to specify alphabetical exten patterns in the dialplans on Asterisk?
All my users would have alpha/numerical ids. I don't want to add a line
for every user in my dialplans.
I searched around, but couldn't
On Thu, Sep 06, 2007 at 02:07:28AM -0600, Al lists wrote:
Hi list,
I'm trying to get some ideas on this subject.
Normally astersik sends emails with voicemail attached trough local MTA.
As far as i know there is no way for asterisk to authenticate to an external
mailserver to relay these
In the capi.conf file, there is a bridge option that allow to native
bridging (CAPI line interconnect) if available, and I found this in the
capi-user mailing list :
I suggest you put bridge=yes into each interface.
Then, when Asterisk bridges two channels, it
Am Donnerstag, den 06.09.2007, 02:07 -0600 schrieb Al lists:
Hi list,
I'm trying to get some ideas on this subject.
Normally astersik sends emails with voicemail attached trough local
MTA.
As far as i know there is no way for asterisk to authenticate to an
external mailserver to relay these
Thank for suggestion now i have done it and it is working fine
One thing i have find many document but i was confuse thats why i have put it
on mailing list if u have or anybody have problem then i m sorry for that.
Anselm Martin Hoffmeister [EMAIL PROTECTED] wrote: Am Mittwoch, den
What's the current thinking on running Asterisk in a UML environment? I
saw some discussion about Xen and asterisk on a Xen DomU.
I'm currently running Asterisk in a UML and have noticed poorer quality
on calls. I'm only using SIP and IAX2 trunks. No hardware adapters. I
guess timing is
On Thu, 6 Sep 2007, lemmel lemmel wrote:
In the capi.conf file, there is a bridge option that allow to native
bridging (CAPI line interconnect) if available, and I found this in the
capi-user mailing list :
I suggest you put bridge=yes into each interface.
Anselm Martin Hoffmeister wrote:
Am Donnerstag, den 06.09.2007, 02:07 -0600 schrieb Al lists:
Hi list,
I'm trying to get some ideas on this subject.
Normally astersik sends emails with voicemail attached trough local
MTA.
As far as i know there is no way for asterisk to authenticate to
Benoit Panizzon wrote:
Hi there
I europe alaw is usual. I have a SIP Phone which perferes ulaw.
When my * box has to transcode alaw to ulaw the sound get's one way choppy.
(alaw = ulaw is choppy, ulaw = alaw is fine).
I managed to fix the issue by forcing my SIP phone to use alaw only,
Hi all,
We want to offer hosted PBX services to some of our clients (maybe
10-20) and were wondering if it makes sense to get a software package
capable of handling multiple virtual tenants or if we should just
create multiple virtual machines in our server each running a single-
tenant
[EMAIL PROTECTED] wrote:
Hi all,
We want to offer hosted PBX services to some of our clients (maybe
10-20) and were wondering if it makes sense to get a software package
capable of handling multiple virtual tenants or if we should just
create multiple virtual machines in our server
Thanks for the prompt response. I apologize if my message came the
wrong way. The objective of my message was to know whether anyone
used multiple instances of asterisk (10-20) within virtual machines
and how well it behaves under that scenario. I know there are many
people using
On Thu, Sep 06, 2007 at 07:30:57AM -0400, Steve Totaro wrote:
I use http://www.dnsexit.com/Direct.sv?cmd=mailRelay to get around port
25 blockage at home and also avoid going into the spam blackhole. It
has an option for no authentication if coming from a defined IP
address. That gets
Thanks for your quick answer :-).
I am a rookie in all this telephony problem, so I'll try to be verbose.
This function does work well. But it works if your ISDN card/driver
supports
it only.
I currently have a Diva Server 4BRI Rev 2, and it seems that there is DSP on
it (voice detection, and
Thanks. Will check that out.
Joseph
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tony
Mountifield
Sent: Wednesday, September 05, 2007 2:38 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] DTMF Relay Problems
In article [EMAIL
On Sat, 1 Sep 2007, Jay R. Ashworth wrote:
On Sun, Sep 02, 2007 at 04:38:19AM +0300, Tzafrir Cohen wrote:
You mentioned that the two disks are identical. Hence there's a large
chance that they're from the same batch. This increases the chance of
them failing together :-p
In practice,
Hi list,
I have a problem with 2 or 3 specific clients.
In the 6 minute, the voip client hear the other one, but the other side can't
hear. After 30 seconds, the both sides recover the audio. And in the
asterisk i have the next notice
will not be disconnected in 31 seconds because it is
On Thu, 6 Sep 2007, lemmel lemmel wrote:
Thanks for your quick answer :-).
I am a rookie in all this telephony problem, so I'll try to be verbose.
This function does work well. But it works if your ISDN card/driver
supports
it only.
I currently have a Diva Server 4BRI Rev 2, and it
Hello, I'm working with our SIP provider to nail down some call quality issues
we're having, and they've asked me to provide SIP debug log files from our
asterisk server. Is there a way to make asterisk 1.4 output only SIP
debugging to a specific log file? Or it is best just to use tcpdump?
Hello Guys,
I am unable to make calls to outside number from some of my extensions.
Internally I am able to make and receive calls between extensions and also I
am able to receive call from outside number. Any suggestions?
Then in am thinking of getting rid of Sysmaster and configure Trixbox
On Thu, 2007-09-06 at 09:58 -0400, Jason Martin wrote:
Hello, I'm working with our SIP provider to nail down some call quality
issues
we're having, and they've asked me to provide SIP debug log files from our
asterisk server. Is there a way to make asterisk 1.4 output only SIP
debugging
[RESOLVED]
Hello Andrew and thx you for your response, which led me to the solution.
You are right concerning the Ringing() and Answer(), so I put this out of my
dialplan.
The way to test with a std phone is a good idea, and permit me to hear the
spandsp CED tone. Very easy to do, and I'm still
Hello
my current test is to send a fax to an analog fax (# is 5656) using the
Asterisk spooler.
I created this file which I copied into /var/spool/asterisk/outgoing
directory:
# fax_out.call
Channel: Zap/4/5656
MaxRetries: 50
RetryTime: 60
WaitTime: 45
Application: txfax
Data:
So please provide a log with
set verbose 5
capi debug
Well, let's go for the debug log :-).
In this capture, I used only one adapter, and I performed a call to
asterisk, which dialed a number (as described in the previous mail) ; so
this is the log :
-- Executing [EMAIL PROTECTED]:2]
I have multiple upstreams in my office. The primary upstream is having some
issues with latency\jitter. I want to move the VoIP traffic to another
interface.
I have the router set to send all traffic destined for local networks out the
respective interfaces. Traffic destined to the Internet
FRIDAY September 7th at 12:30 PM EDT
http://www.asteriskusersconference.org for more information on how to
listen, talk, or both :)
This week, ENUM is the main subject, although our friends at e164.org
haven't been able to talk to us as planned. Come on by and share what
you know about ENUM or
I justed wanted to send out a quick note and remind the Asterisk community about
the Digium Innovation awards, as the dealine for submission is
approaching. (Submissions are due by October 1st.) This is a great
chance to show off your innovative Asterisk-based solutions, and to get
recognized by
Any opinions/comments/recommendations? Before anyone recommends just
buying the virtual PBX service from someone else, we _really_ want to
do this in-house :)
I am all for using VPS (virtual private servers) instead of the classic
multitenant.
Here are some reasons:
- a VPS provides a Linux
On Thu, 6 Sep 2007, lemmel lemmel wrote:
So please provide a log with
set verbose 5
capi debug
Well, let's go for the debug log :-).
In this capture, I used only one adapter, and I performed a call to
asterisk, which dialed a number (as described in the previous mail) ; so
this is
We've just received a bill from bt where it claims that we are making
numerous calls to the same number time after time.
e.g.
01226xx Barnsley20/06/2007 211516:00:00
01226xx Barnsley20/06/2007 121908:55:32
01226xx Barnsley
Maybe you want to provide a full log (including call of the first channel
and the hangup) to my personal mail?
I just added an attachment to this mail.
Thanks a lot :-)
P.S.: this log was generated with verbosity to 5 and with capi debug
Has anybody ever integrated Skype with Asterisk? If you have, which
software would you recommend to accomplish such a task? ChanSkype? And how
reliable are the calls? Did the DTMF tones work? Thanks in advance.
_
Discover
Mojo with Horan Company, LLC wrote:
Just to be clear, I thought that dialtone provision didn't require the
power cable, just generating ring voltages? Can anyone say?
The DC-DC converter on the FXS modules supplies both ringing voltage and
line voltage. If the power connector is not plugged
On Thu, 6 Sep 2007, lemmel lemmel wrote:
Maybe you want to provide a full log (including call of the first channel
and the hangup) to my personal mail?
I just added an attachment to this mail.
Thanks a lot :-)
P.S.: this log was generated with verbosity to 5 and with capi debug
Your
On Thu, Sep 06, 2007 at 04:38:40PM +0100, Senad Jordanovic wrote:
Any opinions/comments/recommendations? Before anyone recommends just
buying the virtual PBX service from someone else, we _really_ want to
do this in-house :)
I am all for using VPS (virtual private servers) instead of the
Hello All,
Does anyone knows a good carrier who can pass DTMF tone while doing Call
Back? Currently, the Call Back system works within US, but as soon as
international users tries to enter phone number the system does not
understand the tones.
I tried to change the sip config to inband, auto,
I am using a2billing as calling card platform with asterisk 1.2.17.
After running for several days, if I issue 'sip show channels' command, I got a
lot of dead sip channels although 'show channels' command only show 5
channels. What cause these dead channels? How can I clean out these dead
Tzafrir Cohen wrote:
On Thu, Sep 06, 2007 at 04:38:40PM +0100, Senad Jordanovic wrote:
Any opinions/comments/recommendations? Before anyone recommends just
buying the virtual PBX service from someone else, we _really_ want
to do this in-house :)
I am all for using VPS (virtual private
A question. are the clients going to be able to manage the PBX? or
are you going to give them the PBX service without access to each
server?
On 9/6/07, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Thu, Sep 06, 2007 at 04:38:40PM +0100, Senad Jordanovic wrote:
Any
Anybody uses Asterisk Java to register an extension to an Asterisk
server ???
Is there any solution for this ?
Phan Anh Vu
DT12.K49.HUT
RDLab ( C9.410 ) HUT
-
Be a better Globetrotter. Get better travel answers from someone who knows.
Yahoo! Answers
Edgar Guadamuz wrote:
A question. are the clients going to be able to manage the PBX?
Yes...
or are you going to give them the PBX service without access to each
server?
Up to you...
Senad
___
Sign up now for AstriCon 2007! September
Your Dial string has errors:
CAPI/contr1/b:103||tT
b: sets the caller number to 'b'. I think
what you are trying to do is
CAPI/contr1/103/b
I just checked the README, and I saw this, thanks :-) (the docs I readed are a
bit old I suppose -e.g.
Eric ManxPower Wieling wrote:
The correct term for this tone is howler. I'm surprised it is not in
indications.conf
I recall seeing it there once, but I'm reaching into the dusty recesses
of my memory right now.
I noticed that all the replies to the OP assumed a SIP handset. The
howler
Hi! - Trying a repost, my first message didn't seem to make the list.
I have one main queue with agents that take calls to our main
phonenumber. Now i want to cascade calls through to the fallback queue
immediately when all the agents in the first queue are 'unreachable' in
any way (be it
I've been working on this the past few days and thought I would put it out
there to see if anyone else has interest in it. It really has nothing to do
with the Digium appliance, I've just been looking for some mass produced
solid state hardware to run small branch offices off of for awhile now
I have an issue with receiving inbound calls.
I've got bandwidth.com trunks incoming to my asterisk box, bandwidth sends all
incoming traffic to one of two IP addresses, and requires outbound traffic go
to either of the same two IP addresses.
I've got to use fromuser=DID on outgoing calls so
I think the testing frame work includes both the components
and system testing.
I wish to add some more test even though all giants may aware,
since i wish to do some contribution to asterisk what ever i can.
i am plannig for the framework and addon as given below, expecting
techies advise in
Jeremy P wrote:
Basically I've taken an HP thin client workstation which is all solid
state and loaded Debian and Asterisk on it (well, Asterisk-GUI too,
but just to prove I could make it appliance-worthy). I'd be
interested in any feedback on how to improve it, specifically on how
to
Debian GNU/Linux 3.1 (Sarge).
This version supports udev 0.056-3 , but it is not installed as a normal
part of the setup process.
Which is my problem...probably. Now I have to figure how to set this up.
Craig
smime.p7s
Description: S/MIME cryptographic signature
We have a Asterisk box acting as a voicemail system and greeting/
call director for our phone system (NEC system). The problem we are
having is that randomly (though most especially with cell phones)
asterisk thinks it is getting a double digit. For example, somebody
will enter
On 9/6/07, Jeremy P [EMAIL PROTECTED] wrote:
I've been working on this the past few days and thought I would put it out
there to see if anyone else has interest in it. It really has nothing to do
with the Digium appliance, I've just been looking for some mass produced
solid state hardware to
On Thu, Sep 06, 2007 at 12:48:44PM -0700, Markham, Craig (FRTC Contractor)
wrote:
Debian GNU/Linux 3.1 (Sarge).
This version supports udev 0.056-3 , but it is not installed as a normal
part of the setup process.
Which is my problem...probably. Now I have to figure how to set this up.
| app_queue.c: No one is answering queue '511' (7/2/0)
The 7/2/0 indicates that you have 7 members in your queue and 2 are
busy. This would indicate that even though those 2 members are busy,
there are still 5 more available members for taking calls. Since there
are available members,
On Thu, Sep 06, 2007 at 01:05:28PM -0600, Jeremy P wrote:
I've been working on this the past few days and thought I would put it out
there to see if anyone else has interest in it. It really has nothing to do
with the Digium appliance, I've just been looking for some mass produced
solid state
To: Twin Cities Asterisk Users
From: [EMAIL PROTECTED]
Subject: TwinCities Asterisk Users Group Meeting this Saturday - Only 1
and 1/2 days away!
Meeting Start: 09/08/2007 - 11:30am
Hello all Twin Cities Asterisk Users,
It's time once again to have another meeting.
I've not had much time to
Greetings list,
I've been asked by someone to help them set up a SIP link between an asterisk
system and an Alcatel OmniPCX (v6 software). The asterisk bit's fine, but I
know nothing about the Alcatel except that it does apparently allow the setup
of SIP trunks.
Does anyone have experience
hi:
i am new to asterisk and dundi. we have some branch office which
will use asterisk in the future. they will form a full-mesh structure
so every site can contact each other directly. i want to try setup
dundi, then we don't need to modify every pbx when a new site add in
the cloud.
Kai-Uwe Jensen wrote:
How are you playing the voice? Do you use something like app_swift
or app_cepstral? Just fixed app_swift for my own installation by
changing the framesize constant definition from 160*4 to 20,
after googling for a similar issue. Works like a charm now. It only
broke
Sure. Sorry to be unclear about it. I was using app_swift-2.0rc1, from
http://www.mezzo.net/asterisk/app_swift.html. Part of that package is
app_swift.c. At line 68, I changed the declaration
const int framesize = 160*4;
to
const int framesize = 20;
That fixed things here. As it seems, that
On Thu, 6 Sep 2007, Jeremy P wrote:
I've been working on this the past few days and thought I would put it out
there to see if anyone else has interest in it. It really has nothing to do
with the Digium appliance, I've just been looking for some mass produced
solid state hardware to run
I've been working on this the past few days and thought I would put it out
there to see if anyone else has interest in it. It really has nothing to do
with the Digium appliance, I've just been looking for some mass produced
solid state hardware to run small branch offices off of for awhile
I know there are many
people using single-tenant and multi-tenant versions of asterisk
management and billing packages, but I don't really know if anyone is
using it within virtual machines and how well that scales.
We have a few FreePBX setups running in virtual machines in environments
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