On Tue, 4 Dec 2007, Vincent wrote:
Does someone know why the posts from some users on Usenet are just one
long line, with no carriage return?
It's called flowed text and defined in RFC 3676. Essentially it lets
people with different width screens accomodate paragraphs of text which is
Hello users,
Actually i wanted to implement Text-To-Speech engine
from cepstral voice using swift application
i tried the documentation of doing this and i was unsuccessful
at doing this work with asterisk
can anybody please help me out finding the solution to installation
thanks in advacnce
Hi All;
I installed one digium card of 2 fxo and 2 fxs, but
the following problems related to the voice are
happening:
1) Sometimes when I call to the PBX, I hear like modem
sound and after little it disapear.
2) There is a disturbance in the background (like the
channel radio disturbance that
Hi Bie,
You have a problem with the postgresql conexion. You are using Realtime?
[Dec 5 07:39:59] ERROR[2342] res_config_pgsql.c: Postgresql RealTime: Failed
to connect database server asterisk on 127.0.0.1. Check debug for more info.
[Dec 5 07:39:59] DEBUG[2342] res_config_pgsql.c:
Hi All;
It is digium analoge card (2 fxo and 2 fxs), so what
do I need to use? And where I can find a link for
that?
Also, is it possible to have a difference voice
volumes to be used each for each Trunk or each user?
Your kindly help is high appreciated.
Regards
Bilal
bilal ghayyad wrote:
Hello,
There is something wrong when using the version 3.0 of X-Lite.
When X-Lite sends INVITE, Asterisk replies OK. And it seems, at first
sight, that Asterisk ignores the ACK signal sent by X-Lite. There's
after a series of Retransmitting of the OK signal, the ACK signals
are well received
Alex Balashov wrote:
I'm sure this has been asked a million times before, but is there an easy
wa to have Asterisk register more than one (distinct) contact binding
concurrently?
The goal is to have two phones register with the same credentials from
different locations and consistently and
Philip Prindeville wrote on Tuesday, 04 December 2007 at 11:58 PM:
Steve Edwards wrote:
On Tue, 4 Dec 2007, Philip Prindeville wrote:
I wanted to write a popcorn app for myself, both to learn how to
script in
Just out of curiosity, what does this have to do with popcorn?
Thanks in
I am sending a call down a EM wink trunk to a adtran tsu600
channelbank. The extension is setup like so...
exten=799179,1,Dial(zap/g2,20,D(9179))
exten=799179,2,Hangup()
It should Dial the Adtran and send some DTMF signals to a telephone on
an fxs module in the Adtran.
Asterisk is seeing the call
Hi,
I use SIP-Realtime to store my SIP-users and I keep the informations
about the SIP-Providers my Asterisk registers to in sip.conf.
I'm running into the following problem. If I set rtcachefriends=yes
because I want to use MWI and run a sip reload because I changed
something in sip.conf,
SIP wrote:
Steve Totaro wrote:
Alex Balashov wrote:
I'm sure this has been asked a million times before, but is there an easy
wa to have Asterisk register more than one (distinct) contact binding
concurrently?
The goal is to have two phones register with the same credentials
Ryan Burke wrote:
I just was looking over the app_waitutil.c and am confused you add 500
to tv.tv_usec on the line msec = (future - tv.tv_sec) * 1000 -
((tv.tv_usec + 500) / 1000);?
Without having looked at Philips code at all, that looks like he is
rounding up?
/Per Jessen, Zürich
--
Where bandwidth is not an issue but good call quality is, is there any
theoretical quality improvement to be had by using slin as the codec
over an inter-Asterisk IAX trunk rather than a-law (or u-law in the US).
Does anyone know what the slin bandwidth is (compared to 64 kbps a-law).
Thanks
Jordan Novak wrote:
I am sending a call down a EM wink trunk to a adtran tsu600
channelbank. The extension is setup like so...
exten=799179,1,Dial(zap/g2,20,D(9179))
exten=799179,2,Hangup()
It should Dial the Adtran and send some DTMF signals to a telephone on
an fxs module in the Adtran.
Well, setting up queues for every user is one option, but it's
troublesome.
Also, nearly all commercial VoIP origination platforms I've seen,
including that of a former Vonage-like employer, support concurrent
contacts in their registrar.
I guess to really do this as a matter of
On Wed, 5 Dec 2007, Trevor G. Hammonds wrote:
As of 19th September 2007, ATT discontinued the service due to the
unavailability of parts for the 1960s-era Audichron equipment, and
declining use of the service.
I don't believe for a minute that it was discontinued due to lack of
parts. I
El Mie, 5 de Diciembre de 2007, 11:45, Michael Melia Jr. escribió:
Does anyone know how I could integrate my Asterisk setup with Outlook so
that when I click on a phone number is my outlook address book it will
dial the number and ring my SIP phone so that I can just pick it up? I
am
I'd look at a program called Outcall. I believe this will handle
everything you'll need.
Michael Melia Jr. wrote:
Does anyone know how I could integrate my Asterisk setup with Outlook
so that when I click on a phone number is my outlook address book it
will dial the number and ring my SIP
Patricio Valarezo Lozano wrote:
Hi, I hope someone could help me, i have a x100p interface for testing
purpose and on each incomming call I redirect the call to handytone 388
atas, the problem comes when i'm during a call and another call comes
in, i hear the call waiting beep (comming
Sam,
Thank you for the suggestion. That is pretty much what I ended up doing
for myself anyway; the real issue is standardising it and doing it on a
mass scale for all users of a platform.
-- Alex
On Wed, 5 Dec 2007, Lutgring, Sam wrote:
Alex;
I would suggest simply registering them as
On Wed, Dec 05, 2007 at 11:07:01AM -0500, SIP wrote:
IM is one of those few scenarios where I think that I'd NOT want to have
possibly multiple logins at the same time. The last thing I need is to
have one half of a conversation on a random machine that I forgot to log
out of -- if nothing
On Wed, Dec 05, 2007 at 12:26:46PM -0500, Jared Smith wrote:
On Wed, 2007-12-05 at 11:45 -0500, Michael Melia Jr. wrote:
Does anyone know how I could integrate my Asterisk setup with Outlook
One of the more popular ones seems to be Outcall, which is now
open-source and available from
Tony Mountifield wrote:
In article [EMAIL PROTECTED],
Ryan Burke [EMAIL PROTECTED] wrote:
I just was looking over the app_waitutil.c and am confused you add 500 to
tv.tv_usec on the line msec = (future - tv.tv_sec) * 1000 - ((tv.tv_usec
+ 500) / 1000);?
It's just doing a standard
Hello,
I'm using Asterisk 1.4.14, and I've noticed that the emails that are sent out
when a user gets a voicemail don't have the timezone set in the header, so
they're appearing in the user's email clients at the wrong time. Has anyone
else seen this? I didn't find any bug reports or other
Hi,
I have a problem with a TDM400P card configuration. Incoming calls are
answered by asterisk, asterisk place the call on the destination
ATA/analog-phone, the phone begins to ring and when our recepcionist pickup
the phone to play a welcome message, she nothing hear on the line during
five or
I would recommend Activa TSP as I prefer its Outook integration than
Outcall's one :
- you're not limited to local contact folders,
- it doesn't need to import contacts
- GUI is simple.
It's based on TAPI and AMI.
A bug in AstManProxy prevent it to be used with it.
When you pick a Contact in
Bill Hackensack wrote:
On Dec 2, 2007 3:42 PM, Carlos Rojas [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
I'm looking the softswitch in digium website, anyone test the
softswitch?
Nope. No one has tested it or used it. Try the one at cisco.com
http://cisco.com.
Digium
Steve Totaro wrote:
SIP wrote:
Steve Totaro wrote:
Alex Balashov wrote:
I'm sure this has been asked a million times before, but is there an easy
wa to have Asterisk register more than one (distinct) contact binding
concurrently?
The goal is to have two
Jason Martin wrote:
Hello,
I'm using Asterisk 1.4.14, and I've noticed that the emails that are sent out
when a user gets a voicemail don't have the timezone set in the header, so
they're appearing in the user's email clients at the wrong time. Has anyone
else seen this? I didn't find
On Wed, 2007-12-05 at 11:45 -0500, Michael Melia Jr. wrote:
Does anyone know how I could integrate my Asterisk setup with Outlook
One of the more popular ones seems to be Outcall, which is now
open-source and available from http://outcall.sourceforge.net. I
haven't tried it personally, so your
On Wednesday 05 December 2007 01:25:19 pm Jason Martin wrote:
Hello,
I'm using Asterisk 1.4.14, and I've noticed that the emails that are sent
out when a user gets a voicemail don't have the timezone set in the header,
so they're appearing in the user's email clients at the wrong time. Has
Philip Prindeville wrote:
Steve Edwards wrote:
On Tue, 4 Dec 2007, Philip Prindeville wrote:
I wanted to write a popcorn app for myself, both to learn how to script in
Just out of curiosity, what does this have to do with popcorn?
Thanks in advance,
Does anyone know how I could integrate my Asterisk setup with Outlook so
that when I click on a phone number is my outlook address book it will
dial the number and ring my SIP phone so that I can just pick it up? I
am interested in this integration for WinXP with Outlook 2003 and
WInVista with
Alex;
I would suggest simply registering them as separate or unique phones and
then ringing multiple phones from the same extension using the .
This way both phones will ring and you can answer based on which one is
local to you. I do this with my desk phone and my X-lite soft phone.
Here is
Ira wrote:
At 11:58 PM 12/4/2007, you wrote:
You used to be able to dial popcorn (767-2676) in any area code (at
least prior to 1982) and get the current time.
I thought it was UL3-2121 when I was younger and occasionally if that
was the only number in the UL3 prefix, dialing
At 11:58 PM 12/4/2007, you wrote:
You used to be able to dial popcorn (767-2676) in any area code (at
least prior to 1982) and get the current time.
I thought it was UL3-2121 when I was younger and occasionally if that
was the only number in the UL3 prefix, dialing just UL3 was enough to
get
We're using 184 here (aka TOS 5/EF).
Not set by iptables though, instead it is set in sip.conf
(tos_sip/tos_audio) and on our Polycom/Cisco phones.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve
Johnson
Sent: Wednesday, December 05, 2007 12:49
To:
At 03:13 12/5/2007, srinivas Antarvedi wrote:
Hello users,
Actually i wanted to implement Text-To-Speech engine
from cepstral voice using swift application
i tried the documentation of doing this and i was unsuccessful
at doing this work with asterisk
can anybody please help me out finding the
Can anyone comment on the DSCP quality of service settings on your
Asterisk server?
The network we're setting up has data on the default VLAN, Asterisk
server and phones on VLAN 4, and we're using Polycom phones with a PC
hooked up to the phone's pass-thru port.
What iptables settings are you
Yes, it is in queues but there isn't anywhere to move them :)
Instead we went ahead and generated whitenoise files just above the
silence supression threshold to use as an alternate which is a little
easier on the ears.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]
Thanks, Darryl,
To clarify:
in /etc/asterisk/sip.conf you have the lines:
tos_sip=cs3; Sets TOS for SIP packets.
tos_audio=ef ; Sets TOS for RTP audio packets.
and in your Polycom configuration [I'm using Polycom's sip 2.2.0] you
have something like (this
On Wednesday 05 December 2007 01:25:19 pm Jason Martin wrote:
Hello,
I'm using Asterisk 1.4.14, and I've noticed that the emails that are sent
out when a user gets a voicemail don't have the timezone set in the header,
so they're appearing in the user's email clients at the wrong time. Has
Is it possible for a Cisco 7960 phone with SIP firmware to connect to 2
different SIP servers @ the same time?
I currently have an asterisk box @ home with several sip extensions and
a Nortel C15k phoneswitch at work (not the pbx, the full phone switch).
I can connect from the SIP phone to the
Looks fine to me, you only need to specify DSCP or TOS (may need to
check the manual for which it takes first).
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve
Johnson
Sent: Wednesday, December 05, 2007 14:02
To: asterisk-users@lists.digium.com
Thanks for the suggestions so far. I don't like the idea that I have to
give full control with OutCall but it seems to be the case with most of
the solutions out there. I have downloaded and tested OutCall on
Windows Vista and Outlook 2007. It doesn't seem to work 100% with
Outlook 2007.
Mojo with Horan Company, LLC wrote:
Patricio Valarezo Lozano wrote:
Hi, I hope someone could help me, i have a x100p interface for testing
purpose and on each incomming call I redirect the call to handytone 388
atas, the problem comes when i'm during a call and another call comes
in, i
SIP wrote:
Every machine in a in a Windows environment must be configured to join a
domain. A user must also be setup in that domain to log in. It is more
secure to lock that user into a single login session so that if they are
logged in at one machine, they cannot login somewhere else.
In article [EMAIL PROTECTED],
Ryan Burke [EMAIL PROTECTED] wrote:
I just was looking over the app_waitutil.c and am confused you add 500 to
tv.tv_usec on the line msec = (future - tv.tv_sec) * 1000 - ((tv.tv_usec
+ 500) / 1000);?
It's just doing a standard round to nearest integer division,
Partially answering my own question, it looks like slin is a 128 kbps
codec.
Peter
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Whisker,
Peter
Sent: 05 December 2007 16:02
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
Hi.
I wanted to write a popcorn app for myself, both to learn how to
script in extensions.conf, but also because it was something handy.
Along the way, I found myself doing something like:
[popcorn]
exten = s,1,Set(FUTURETIME=$[${EPOCH} + 10])
...
exten = s,n,While(${EPOCH}
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Good Morning,
My problem was that the context wasn't the same in my voicemail.conf and
in my sip.conf!! One was 'default' and the other 'device'
I have put 'default' everywhere and it's working!
Have a nice day
Jared Smith a écrit :
On Tue,
Does this number (you are dialing) has been ported from a different Telco?
When you dial from the other city and you get service not available you
may be dialing from a different Telco that either has no route aggreement
for the dialed network, or the number portability database (of Out of city
On Wed, 5 Dec 2007, SIP wrote:
I just don't get the whole FUD issue with this. I understand that it's
simply part of the way PBX systems work... but discounting the option as
'dangerous' is just masking the issue.
I would tend to agree. One of the key value propositions proffered by
VoIP
I use SIP-Realtime to store my SIP-users and I keep the informations
about the SIP-Providers my Asterisk registers to in sip.conf.
I'm running into the following problem. If I set rtcachefriends=yes
because I want to use MWI and run a sip reload because I changed
something in sip.conf,
Steve Edwards wrote:
On Wed, 5 Dec 2007, Trevor G. Hammonds wrote:
As of 19th September 2007, ATT discontinued the service due to the
unavailability of parts for the 1960s-era Audichron equipment, and
declining use of the service.
I don't believe for a minute that it was
bilal ghayyad wrote:
1) Sometimes when I call to the PBX, I hear like modem
sound and after little it disapear.
2) There is a disturbance in the background (like the
channel radio disturbance that might happen if the
frequency was not captured well), and that disturbance
appear much more
Hi,
I have the following setup:
(sip clients) -internet- asterisk A -IAX- asterisk B -PRI- (pstn)
This works fine for regular calls sip-pstn. the calls go through
perfectly. However, when one of the sip clients (a snom320) is set to
redirect to the pstn, then all I hear is congestion tones
Hi,
after I fixed my problem with the playback() application, I now have the
next strange one.
When I dial the number of our client, located in another town, I get a
connection to the asterisk server, I can talk to my client or listen to
his mailbox.
If some in the town of this client calls
Ring all queues would be easier I would think.
Thanks,
Steve Totaro
Alex Balashov wrote:
Sam,
Thank you for the suggestion. That is pretty much what I ended up doing
for myself anyway; the real issue is standardising it and doing it on a
mass scale for all users of a platform.
-- Alex
In article
[EMAIL PROTECTED],
Ryan Burke [EMAIL PROTECTED] wrote:
I just was looking over the app_waitutil.c and am confused you add 500
to
tv.tv_usec on the line msec = (future - tv.tv_sec) * 1000 -
((tv.tv_usec
+ 500) / 1000);?
It's just doing a standard round to nearest integer
Steve Totaro wrote:
Alex Balashov wrote:
I'm sure this has been asked a million times before, but is there an easy
wa to have Asterisk register more than one (distinct) contact binding
concurrently?
The goal is to have two phones register with the same credentials from
different
Hi Guillermo,
I am not using Realtime.., why it seems line turned on? .. how to turn it
off?
BTW .. I have put type=friend into my sip.conf ..but the same problem still
occurs (I am unable to register the SPA-3102 ;(
Regards
bie
- Original Message -
From: Guillermo Rodriguez [EMAIL
Hi, I have just configure a Soundpoint 550 to work with Asterisk, it appear
Registered to the asterisk server, and appear in asterisk console with SIP
SHOW PEERS, and can receive calls, but when I try to dial, it launch a tone
as if not line to dial, also can not stream audio to the other end when
There are probably a half dozen or more software apps that can do this.
Most are free last time I checked. Google is your friend.
From: Michael Melia Jr. [mailto:[EMAIL PROTECTED]
Sent: Wednesday, December 05, 2007 8:45 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Asterisk
Snapanumber is the best way to do this.
It's a commercial app so has a license fee but works great.
Cant comment about Outlook2007 but works great with 2003 for me.
Regards,
Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED]
+1-212-203-4357
+61-2-9016-5642 (Sydney in-dial).
Hello!
Please help me with decision problem. I need to organize voip telephony in
office. I have 2 phone lines(2 physical number) for phone and fax.I
need to recive call on 1 phone then redirect it to neccessary phone or
fax. Can Asterisk do that ?
Thanks in advance
Besides grandstream-doorphone transplant surgery, no. But it does have
PoE. It's cheap, especially if you already have a doorphone. If you
used a GXP-2000 you can use the display and it supports XML idle
screens.
On Dec 4, 2007 2:53 AM, Nick Seraphin [EMAIL PROTECTED] wrote:
On a similar
Hi All, as good?
I am trying to make a call for the Unicall channels and after the
exchange of signalling and sending of the digits asterisk locks up
with the sending of the signalling E and the TELCO says that
asterisk would have to send signalling F, as to make for asterisk to
send signalling F?
I have seen a beta-level unit that also supported POE.
With regards to non-beta hardware, standard analog doorphones work
pretty well with Linksys SPA units.
PaulH
On Tue, 2007-12-04 at 02:53 -0500, Nick Seraphin wrote:
On a similar note... has anyone ever seen a SIP-based door intercom
Hello all, as good?
I am trying to use the package astunicall-1.2.21.0.1 with a Sangoma
A104D card and 04 links E1 mfc/r2 in Brazil. The compilation occurred,
normally and links is UP if I place in Loop and I obtain to effect
called in Loop, but when I extend for the PSTN, links reports the
On Dec 5, 2007 10:07 AM, SIP [EMAIL PROTECTED] wrote:
Steve Totaro wrote:
SIP wrote:
Steve Totaro wrote:
Alex Balashov wrote:
I'm sure this has been asked a million times before, but is there an
easy
wa to have Asterisk register more than one (distinct) contact binding
Yes Asterisk can receive the calls and based either on the line the
call is on or some other method route the call to a destination. That
being said there are 2 things to keep in mind, the hardware cost to
setup 2 incoming lines and a analog port for the fax as well as phones
may be high for a 2
Quoting Paul Hales [EMAIL PROTECTED]:
another option is use some sort of linux based device n770, or even an
nslu2, and program a sip client to behave however you like, then just
fit the thing with a usb based speaker/mic.
actually, a gamepad, speaker and mic with an n770 behind a piece of
I am running 1.4.10.1. I have a macro that is called from default for a
certain extension (both below). I added NoCDR to s to try and stop
extra CDR records, but I am still getting them. Any idea how to stop them?
extensions.conf:
[macro-STDEXT]
exten =s,1,NoCDR()
exten
Michael Melia Jr. wrote:
Thanks for the suggestions so far. I don't like the idea that I have to
give full control with OutCall but it seems to be the case with most of
the solutions out there. I have downloaded and tested OutCall on
Windows Vista and Outlook 2007. It doesn't seem to work
On Wed, 5 Dec 2007, Lacy Moore wrote:
the one you are logged into. Same as Asterisk. I can carry a phone
with me, and plug it in and access my Asterisk server. I can login
using softphones. Whatever phone I am on will ring.
Unless the reregistration interval is fairly frequent,
what module does the TDM804B use/need?
Jerry
___
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asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
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I don't think it ever gets that cold here in Australia.
PaulH
On Wed, 2007-12-05 at 21:24 -0500, Jon Pounder wrote:
Quoting Paul Hales [EMAIL PROTECTED]:
another option is use some sort of linux based device n770, or even an
nslu2, and program a sip client to behave however you like,
Lol, not even in Melbourne huh - BTW it's snowing here in NY again tonight :)
Regards,
Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED]
+1-212-203-4357
+61-2-9016-5642 (Sydney in-dial).
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul Hales
Sent:
Why not an ATA that has an FXS port with an autoanswer doorbox like this one:
http://www.vikingelectronics.com/products/view_product.php?pid=428
On Dec 5, 2007 9:48 PM, Paul Hales [EMAIL PROTECTED] wrote:
I don't think it ever gets that cold here in Australia.
PaulH
On Wed, 2007-12-05 at
Hello
Some of our customers call with CID blocked. I'd like to save
those numbers into a SQLite database using a command-line PHP script,
so that I can...
1. Edit the CID name through a PHP web script which will just list all
the customers in the database who have a phone number but no
At 19:19 12/5/2007, Ricardo Melendez wrote:
Hi, I have just configure a Soundpoint 550 to work with Asterisk, it appear
Registered to the asterisk server, and appear in asterisk console with SIP
SHOW PEERS, and can receive calls, but when I try to dial, it launch a tone
as if not line to dial,
Vincent wrote:
exten = 777,n,ExecIf($[${LEN(${CALLERIDNUM})} =
10],AGI(/root/dummy.php),${CALLERIDNUM})
The line break is not a good idea.
It doesn't look like ExecIf() is the right way to have Asterisk run an
AGI script conditionnally. What would be the right way to do this?
Wrong syntax.
On Thu, 06 Dec 2007 05:11:24 +0100, Philipp Kempgen
[EMAIL PROTECTED] wrote:
The line break is not a good idea.
It's not in the script, just my news reader :-)
Not sure about more than one argument. Maybe
Both work. Thanks a lot!
___
--Bandwidth and
Hi!
Steve Johnson wrote:
The network we're setting up has data on the default VLAN, Asterisk
server and phones on VLAN 4, and we're using Polycom phones with a PC
hooked up to the phone's pass-thru port.
If you are using VLAN, than you also look at new options in trunk
cos_sip and cos_audio
New to Asterix and perhaps someone can help.
The plnned configuration is that the Quintums are to register to the Asterix
and the signalling to be handled by the Asterix but the media (G 729 code)
to be directed to the service provider.
Thanks Shaun
Mark,
This is the results
[EMAIL PROTECTED] ~]# cat /proc/cpuinfo
processor : 0
vendor_id : GenuineIntel
cpu family : 15
model : 4
model name : Intel(R) Pentium(R) 4 CPU 3.00GHz
stepping: 1
cpu MHz : 2993.146
cache size : 1024 KB
physical id
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