Hello
I have a
--
Best Regards
Shakeel Abbas
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On 24/08/09 6:27 PM, ABBAS SHAKEEL wrote:
Hello
I have a
:)
hmmm might need a little more info
--
Cheers,
Matt Riddell
Director
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Sorry that was sent by mistake
The question is . I have configured Asterisk with TDM400P i can
recieve calls every thing goes fine...
But one is unclear to me.
If i want to intiate a call to a PSTN number or any mobile number as we
normally do with our phones How can i do
On 24/08/09 6:31 PM, ABBAS SHAKEEL wrote:
Sorry that was sent by mistake
The question is . I have configured Asterisk with TDM400P i can
recieve calls every thing goes fine...
But one is unclear to me.
If i want to intiate a call to a PSTN number or any mobile number as we
Sorry that was sent by mistake i pressed ctrl+s unintentionally
The question is . I have configured Asterisk with TDM400P i can
recieve calls every thing goes fine...
But one is unclear to me.
If i want to intiate a call to a PSTN number or any mobile number as we
normally do
At 11:31 PM 8/23/2009, you wrote:
If i want to intiate a call to a PSTN number or any mobile number as
we normally do with our phones How can i do that with asterisk .
Unless I messed up this uses outgoing DHADI channel 1 to dial 10
digit local numbers with the required 1310 prefix
Thanks Matt and Ira
this worked for me...
exten = 123,2,Dial(DAHDI/2/thephonenumber)
Any body let me know . I want to know if i can say some thing.
It do dail but dont say any thing . ie the bell rings at dailed number
but when the call is picked nothing happens
How can it say some
On 24/08/09 7:31 PM, ABBAS SHAKEEL wrote:
Thanks Matt and Ira
this worked for me...
exten = 123,2,Dial(DAHDI/2/thephonenumber)
Any body let me know . I want to know if i can say some thing.
It do dail but dont say any thing . ie the bell rings at dailed
number but when the call
I am not dailing 123 but i have changed it when shown on forums i just want
to illustrate the dial function :)
Thanks
On Mon, Aug 24, 2009 at 12:37 PM, Matt Riddell li...@venturevoip.comwrote:
On 24/08/09 7:31 PM, ABBAS SHAKEEL wrote:
Thanks Matt and Ira
this worked for me...
exten =
On 24/08/09 7:45 PM, ABBAS SHAKEEL wrote:
I am not dailing 123 but i have changed it when shown on forums i just
want to illustrate the dial function :)
Thanks
Yeah, I just meant to try echo instead of dial - it should echo back to
you what you send to it - that way you can check that your
Thanks Matt for the value able info
Echo dont works for me :(
One more thing what i want to do is making a call to some one and playing a
sound like (I Shakeel is calling you ..if you want to be friend press 1 or
..)
thanks
--
Best Regards
Shakeel Abbas
On 24/08/09 8:07 PM, ABBAS SHAKEEL wrote:
Thanks Matt for the value able info
Echo dont works for me :(
One more thing what i want to do is making a call to some one and
playing a sound like (I Shakeel is calling you ..if you want to be
friend press 1 or ..)
Use the
Hi, im trying to build a UAC and I'm coming up with some trouble whenever I
receive a SIP 491 Request Pending Response. This happens because I try to place
a call on hold using an Invite request rigth before Asterisk sends me a
Re-Invite for the same call. I respond to the 491 response with an
I am also having the same issue with voicemail. I have the exact setup you
have described. Additionally, I am getting segmentation faults during an
asterisk reload. This is happening intermittently.
Below is the core dump backtrace. Not sure how to determine if it an asterisk
or odbc
On 24/08/09 8:54 PM, John Riek wrote:
I am also having the same issue with voicemail. I have the exact setup you
have described. Additionally, I am getting segmentation faults during an
asterisk reload. This is happening intermittently.
Below is the core dump backtrace. Not sure how to
thanks Alot it worked for me
using *G(context^exten^pri)*: If the call is answered, transfer both parties
to the specified context and extension. The calling party is transferred to
priority x, and the called party to priority x+1. This allows the dialplan
to distinguish between the calling and
On 24/08/09 8:54 PM, John Riek wrote:
I am also having the same issue with voicemail. I have the exact setup you
have described. Additionally, I am getting segmentation faults during an
asterisk reload. This is happening intermittently.
Below is the core dump backtrace. Not sure how to
On Fri, 21 Aug 2009, Olivier wrote:
So basically it's harmless, unless you actually have such a card.
Yes, but as you mentioned, most don't have a transcoder card.
My opinion is such message shouldn't be send at all for those environments
where there is no transcoder card, (as it will
Hopefully upgrading to the latest 1.4 version of asterisk will fix it.
Yeah, that'd be the best bet for the moment - if it doesn't you'll need
to open a bug on http://issues.asterisk.org
I'd make a backup of everything before you upgrade - just in case :)
--
Cheers,
Matt Riddell
Director
Hi!
I have the following setup:
PSTN--Asterisk-SIP--Asterisk
GW/LCR \ \ ...
\ \ ...
\ --SIP--Asterisk
\ ...
---Asterisk
The GW-Asterisk just does the
hello,
I tried to compil asterisk-addons-1.6.2.0-rc1,
and I have that error:
[CC] res_config_mysql.c - res_config_mysql.o
res_config_mysql.c:1367: error: unknown field âupdate2_funcâ specified in
initializer
res_config_mysql.c: In function âparse_configâ:
res_config_mysql.c:1432: error:
From voip-info.org :
Asterisk will look for these files in the /var/lib/asterisk/keys
directory, so copy them there and make sure only the asterisk user id
can read the keys and that no one can write over them.
How do I know my Asterisk User ID ??
Jonas.
Hellos,
I am looking for the sounds used in this ivr example
http://www.voip-info.org/wiki/view/Asterisk+Tips+follow+me. The one with
6900.
Any assistance is welcome.
--
Best Regards,
James Mutuku Ndeti
Agile Systems Limited
+254722490994
www.agile.co.ke
mutuku.wordpress.com
Has your
Hi at all !
I've a problem and I don't know how to solve it.
My configuration is the following:
ISDN LINE --- PATTON (SIP) --- ASTERISK
in asterisk my sip.conf for sip patton account is the following:
[general]
port=5060
bindaddr=0.0.0.0
context=default
language=it
limitonpeers=yes
Are you using newest Asterisk versions? There were some similar problems
fixed recently:
https://issues.asterisk.org/view.php?id=13849
https://issues.asterisk.org/view.php?id=14239
https://issues.asterisk.org/view.php?id=14584
regards
klaus
Guillén Melo, Joaquin schrieb:
Hi, im trying to build
Hi,
I want to show the QueueName to my Queue Member.
I try to find the solution to show the QueueName near the callerid of
each call.
Can somebody help me ?
Thanks
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-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Thalassoline - Service technique wrote:
I want to show the QueueName to my Queue Member.
I try to find the solution to show the QueueName near the callerid of
each call.
Can somebody help me ?
Set the CALLERID(name) variable prior to sending
Since the tutorial is 4+ years old and the Wiki author wasn't nice enough to
include the source for /var/lib/asterisk/sounds/portable-number-ivr, the
simplest solution I can offer is to either record these sounds yourself or
to install swift and pipe out these files per the quoted values. Since
Hi everybody,
I'm trying my Asterisk to send emails when a new message arribes to a
voicemail user but no email arribes.
my voicemail configuration is the following:
VOICEMAIL.CONF:
[general]
format=wav
serveremail=aster...@mydomain.com
attach=yes
maxmsg=20
maxsecs=180
minsecs=3
maxsilence=10
I have a small script that do the trick for you.
At you terminal use the follow
wget
http://www.fastway.com.br/wp-content/uploads/2009/02/install_asterisk.sh
chmod 755 install_asterisk.sh
sudo ./install_asterisk.sh
Valter
2009/8/21 aster...@opensourcesolution.in
hello friends,
i have to
Start with simple mail testing (forget asterisk)
Does mx1.datagrama.net accept messages for testu...@mydomain.com ? Try a
telnet session first...
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joan Antoni
Terre
Sent:
Hi
Has anyone already use asterisk 1.6.1.1 with asterisk GUI last release ?
I'm trying it but I have this problem :
Just after I logged, I have system status main page but no other links where
I can click to go to other pages (remember left panel!)
Regards
Harry
Make sure regular sendmail is working; asterisk voicemail uses this unless
you reconfigure it.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michelle
Dupuis
Sent: Monday, August 24, 2009 9:09 AM
To: 'Asterisk Users List'
Hi Michelle,
If I try telnet mx1.datagrama.net
I have no answer, I get:
Trying 212.9.65.110...
¿?
2009/8/24 Michelle Dupuis supp...@ocg.ca
Start with simple mail testing (forget asterisk)
Does mx1.datagrama.net accept messages for testu...@mydomain.com ? Try a
telnet session first...
2009/8/24 Danny Nicholas da...@debsinc.com
Danny,
sorry if it's a silly question but how can I check if sendmail is working'
Thanks
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On Mon, Aug 24, 2009 at 9:25 AM, Joan Antoni Terrenebh...@gmail.com wrote:
Hi Michelle,
If I try telnet mx1.datagrama.net
I have no answer, I get:
Trying 212.9.65.110...
¿?
telnet mx1.datagrama.net 25
that's a space, then the port, in this case, 25.
is ms1.datagrama.net what you really
You also need to specify the port so telnet mx1.datagrama.net 25 return
is the command to use.
db
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joan Antoni
Terre
Sent: Monday, August 24, 2009 10:25 AM
To: Asterisk Users Mailing
Hi Jonas
Type 'id asterisk' at your command line.
It should return uid gid and all groups the asterisk user belongs to.
Cheers
Bails
jonas kellens wrote:
From voip-info.org :
/Asterisk will look for these files in the /var/lib/asterisk/keys
directory, so copy them there and make sure only
Danny,
I've done as you explained and get the same messages at /var/log/maillog
What I've seen at /var/spool/mail/root the following
Subject: Returned mail: see transcript for details
Auto-Submitted: auto-generated (failure)
This is a MIME-encapsulated message
Not silly at all. Voicemail.conf specifies that /usr/bin/sendmail -t will
be used to send the mail, so doing this command:
Sendmail -t To: nebh...@gmail.com followed by enter and ctrl-d should pop
an empty email from root into your inbox if all is well.
_
From:
Jonathan,
now I've done telnet mx1.datagrama.net 25
And I've got:
Trying 212.9.65.110...
Connected to mx1.datagrama.net (212.9.65.110).
Escape character is '^]'.
220 mailhub03.datagrama.net ESMTP Datagrama
It looks as it has connected but has not asked for any user / Password
The receiving server does not ask for any user id or password. The
protocal says, the sender has to just send the user or pass command with
the data required.
Try reading /var/log/mail(if you have access), at least that's where the
outgoing mail logs on my servers are.
Lyle
Joan Antoni Terre
Danny,
my sendmail.cf is not at /etc but at /etc/mail.
These lines you mention are not at the end of this file. Enclosed you'll
find my /etc/mail/sendmail.cf
Many thanks
2009/8/24 Danny Nicholas da...@debsinc.com
In /etc/sendmail.cf, do you have these lines at the end?
Tuser Asterisk
harry R wrote:
Hi
Has anyone already use asterisk 1.6.1.1 with asterisk GUI last release ?
I'm trying it but I have this problem :
Just after I logged, I have system status main page but no other links
where I can click to go to other pages (remember left panel!)
I have Asterisk 1.6.1.4 and
In /etc/sendmail.cf, do you have these lines at the end?
Tuser Asterisk
Tuser asterisk
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joan Antoni
Terre
Sent: Monday, August 24, 2009 9:59 AM
To: Asterisk Users Mailing
On Monday 24 August 2009 04:44:18 am Matt Riddell wrote:
Hopefully upgrading to the latest 1.4 version of asterisk will fix it.
Yeah, that'd be the best bet for the moment - if it doesn't you'll need
to open a bug on http://issues.asterisk.org
I'd make a backup of everything before you
Bails,
thanks for your reply.
bash-3.2# id asterisk
id: asterisk: No such user
So how do I know my Asterisk User ID ??
Greetingz,
Jonas.
On Mon, 2009-08-24 at 15:49 +0100, bails wrote:
Hi Jonas
Type 'id asterisk' at your command line.
It should return uid gid and all groups the asterisk
On Monday 24 August 2009 05:17:24 am BERGANZ François wrote:
I tried to compil asterisk-addons-1.6.2.0-rc1,
and I have that error:
[CC] res_config_mysql.c - res_config_mysql.o
res_config_mysql.c:1367: error: unknown field âupdate2_funcâ specified in
initializer
res_config_mysql.c: In
This one is getting a little over my pay grade. I'd try adding the two
lines at the end and restarting the mail daemon.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joan Antoni
Terre
Sent: Monday, August 24, 2009 10:21
2009/8/21 aster...@opensourcesolution.in
i have to configures asterisk n my hardware details are
Is it just me, or would you think someone from a domain named like
Open Source Solution should be able to figure this one out...
On Mon, 24 Aug 2009, Valter Nogueira wrote:
I have a small script
Hi Danny,
I've tryed it but still the same. It looks that my ISP SMTP server doesn't
like the sender, which I guess is r...@localhost.localdomain. Do you know
how to change this sender?
Many thanks
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Un-top-posting...
jonas kellens wrote:
How do I know my Asterisk User ID ??
On Mon, 24 Aug 2009, bails wrote:
Type 'id asterisk' at your command line. It should return uid gid and
all groups the asterisk user belongs to.
This assumes you have a user named asterisk. Also assumes that
Hi All -
I'm setting up a corporate emergency broadcast system that uses an
autodialer to contact all company employees. Everything works fine
except if the auto-dialed calls go to the end users' voicemail. If
that happens, asterisk starts playback of the emergency message while
the voicemail
That's what the MASQUERADE(localhost.localdomain) is supposed to do; Don't
know why it is not.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joan Antoni
Terre
Sent: Monday, August 24, 2009 10:57 AM
To: Asterisk Users
Hello,
I need help to install a digium card TC400P.
I compiled the dahdi source, but dahdi dont find the card!
debian:/usr/src/dahdi-linux-complete-2.2.0.2+2.2.0# lspci
11:03.0 Ethernet controller: Digium, Inc. Wildcard TC400P transcoder base
card (rev 11)
BERGANZ François wrote:
debian:/usr/src/dahdi-linux-complete-2.2.0.2+2.2.0# /etc/init.d/dahdi
restart
Unloading DAHDI hardware modules: done
Loading DAHDI hardware modules:
wct4xxp: done wcte12xp: done wct1xxp: done wcte11xp: done
wctdm24xxp: done wcfxo: done wctdm: done
What have I to do?
Cordialement,
Pensez à l'Environnement, n'imprimez ce mail que si nécessaire.
-Message d'origine-
De : asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] De la part de Sean Bright
Envoyé : lundi 24 août 2009 18:21
À : Asterisk
I will consider it and to change it to DAHDI too.
It woul be great if repository had files named CURRENT like
asterisk_1.4.CURRENT so would have no need to change any script.
Thanks,
Valter
2009/8/24 Steve Edwards asterisk@sedwards.com
2009/8/21 aster...@opensourcesolution.in
i
If I do #dmesg, I have it :
[ 101.994189] Unregistered codec translator 'DTE Decoder' with 92
transcoders (srcs=0101, dsts=000c)
[ 101.994189] Unregistered codec translator 'DTE Encoder' with 92
transcoders (srcs=000c, dsts=0101)
[ 101.999162] dahdi_transcode:
BERGANZ François wrote:
What have I to do?
Nothing. The system recognizes the card, and the appropriate module is loading.
What is happening that makes you think it isn't working?
--
Sean Bright
sean.bri...@gmail.com
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Do a quick search for SMTP commands - to simulate a complete session via
telnet.
Most MTA's will check sender and recipient for validity, relaying, etc. Be
sure both are reasonable and acceptable to host using telnet first.
If you are new to sendmail.cf, read the instructions at the top of
On Mon, Aug 24, 2009 at 12:49 PM, Sean Bright sean.bri...@gmail.com wrote:
BERGANZ François wrote:
What have I to do?
Nothing. The system recognizes the card, and the appropriate module is
loading.
What is happening that makes you think it isn't working?
--
Sean Bright
Steve Totaro wrote:
No hardware timing source found in /proc/dahdi, loading dahdi_dummy
would make me think it is not loading correctly.
The TC400P is a transcoder card. It is not a timing source.
--
Sean Bright
sean.bri...@gmail.com
___
--
In voicemail.conf you can set serveremail=asterisk so some other
address and this will be used as the sender's email address, at least
as I understand things.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Aug 24, 2009, at 9:50 AM, Michelle Dupuis wrote:
Do a
On Mon, Aug 24, 2009 at 1:06 PM, Sean Bright sean.bri...@gmail.com wrote:
Steve Totaro wrote:
No hardware timing source found in /proc/dahdi, loading dahdi_dummy
would make me think it is not loading correctly.
The TC400P is a transcoder card. It is not a timing source.
--
Sean Bright
Oh my god..
Today its saying there is NOONE to take your call.I am using IdeaSIP
What could be the reasons ?
It was working perfectly till saturday .
On Thu, Aug 20, 2009 at 11:54 PM, David @ULC ucoms2...@gmail.com wrote:
IdeaSIP worked perfect for me.
On Thu, Aug 20, 2009 at 11:27
At 9:04 AM on 24 Aug 2009, Steve Edwards wrote:
Un-top-posting...
jonas kellens wrote:
How do I know my Asterisk User ID ??
On Mon, 24 Aug 2009, bails wrote:
Type 'id asterisk' at your command line. It should return uid gid
and all groups the asterisk user belongs to.
This
For a newbie, this would be preferable;
ps --no-headers -o uname -C asterisk
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of C. Chad
Wallace
Sent: Monday, August 24, 2009 2:13 PM
To:
A quick look at the system shows you're not logged in, which is why
you're getting that message.
N.
David @ULC wrote:
Oh my god..
Today its saying there is NOONE to take your call.I am using IdeaSIP
What could be the reasons ?
It was working perfectly till saturday .
On Thu, Aug 20,
I keep getting a red alarm when trying to setup asterisk to use my
TE420B EC. I only have a blank context setup in my extensions.conf as
I haven't started to config that until I can clear this red alarm. I
don't have physical access to the server, so I can't go reseat the
Hello.
I've been seting up a small VoIP setup, with roughly 5 persons, doing
essentially some Meetme conferences.
People have been experiencing some quality problems with the sound.
Essentially delay, and some tolerable echo.
I'd appreciate advice on how to troubleshoot this issue. What could
Guillaume Yziquel a écrit :
Hello.
I've been seting up a small VoIP setup, with roughly 5 persons, doing
essentially some Meetme conferences.
People have been experiencing some quality problems with the sound.
Essentially delay, and some tolerable echo.
I'd appreciate advice on how
My IPKall number is sending a CLID of a different number!!
Since I was looking for the correct CLID, my Asterisk rejected it, and
it went to, I assume, IPKall asterisk VM!!!
Sounds like IPKall is really wrapped around the axle shaft today!
John Novack
David @ULC wrote:
Oh my god..
Today
you're not logged in means ?
On Mon, Aug 24, 2009 at 11:39 PM, David @ULC ucoms2...@gmail.com wrote:
Oh my god..
Today its saying there is NOONE to take your call.I am using IdeaSIP
What could be the reasons ?
It was working perfectly till saturday .
On Thu, Aug 20, 2009 at
John Novack wrote:
My IPKall number is sending a CLID of a different number!!
Since I was looking for the correct CLID, my Asterisk rejected it, and
it went to, I assume, IPKall asterisk VM!!!
Sounds like IPKall is really wrapped around the axle shaft today!
John Novack
Searching their
Means your username is not registered on the IdeaSIP system (your
client/phone is not logged into IdeaSIP).
N.
David @ULC wrote:
you're not logged in means ?
On Mon, Aug 24, 2009 at 11:39 PM, David @ULC ucoms2...@gmail.com
mailto:ucoms2...@gmail.com wrote:
Oh my god..
On Mon, 2009-08-24 at 17:56 +0200, Joan Antoni Terre wrote:
[]
I've tryed it but still the same. It looks that my ISP SMTP server
doesn't like the sender, which I guess is r...@localhost.localdomain.
Do you know how to change this sender?
Set a sane/correct hostname (which is resolvable
On Mon, 2009-08-24 at 11:12 -0500, Danny Nicholas wrote:
That’s what the MASQUERADE(localhost.localdomain) is supposed to do;
Don’t know why it is not.
It's commented out. The dnl at the begin of the line means delete
'til newline for m4 (which processes that file and produces the .cf
file).
You
Check your hostname settings, hosts file, and order of name resolution...
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bernd
Petrovitsch
Sent: Monday, August 24, 2009 5:26 PM
To: Asterisk Users List
I'd appreciate it if someone could give me an answer to using LDAP in Asterisk
1.6.x
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Klaverstyn
Sent: Thursday, 20 August 2009 4:01 PM
To: 'Asterisk Users Mailing List'
Subject:
Not an LDAP user but perhaps using AGI to access your LDAP server may be
a solution? Looks like it may require some work but I wouldn't think it
to be too hard.
sl
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David
Klaverstyn
trebaum schreef:
I keep getting a red alarm when trying to setup asterisk to use my
TE420B EC. I only have a blank context setup in my extensions.conf as
I haven't started to config that until I can clear this red alarm. I
don't have physical access to the server, so I can't go reseat the
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