Yep make the server with dynamic ip register to server with static ip ( sip or iax both will do but in sip keep nat=yes while making extension ) On 12/11/06,
Rosli Sukri [EMAIL PROTECTED] wrote:
u need another box say box a with real/addressable ip address. create an iax entry in box a and have
Hi all,
I have found that when I use an announcement at the start of the call it
results in a useless answerphone message if the call goes onto
answerphone for any reason - the message being a chopped off version of
the announcement.
Does anyone know of a good way to detect that an
Dear All,
The resolution to the problem below was very easy and I guess that what
made it very hard:
callerid=asreceived
signalling=pri_cpe
switchtype=> euroisdn
context=from-zaptel
group=0
channel=>1-15,17-31
Thx
MAG
"Mohamed A. Gombolaty" wrote:
Dear All,
I have an asterisk server
On 11/12/06, Vicky [EMAIL PROTECTED] wrote:
Yep make the server with dynamic ip register to server with static ip ( sip
or iax both will do but in sip keep nat=yes while making extension )
the problem is that the server with dynamic ip can't register on the
other server!
This is the
Tom Lynn wrote:
Ron,
The guy is trying to help you.
Tom,
I believe it!
Go to the link and read it. There is a feature that you can use to
play a recording into the voice channel. Mine is set so when you
press #9, the caller hears the lots of monkeys recording.
I am not sure if that is
je . a écrit :
Thanks for your response. I'm looking specifically at
Asterisk in a SIP-only implementation. So no need for
Asterisk to transfer calls between PSTN and SIP. Is
there in such a case still a need for a PBX/Asterisk?
As you can see in a typical SIP communication diagram,
Am Sonntag, den 12.11.2006, 08:50 + schrieb Nic Hughes:
Hi all,
I have found that when I use an announcement at the start of the call it
results in a useless answerphone message if the call goes onto
answerphone for any reason - the message being a chopped off version of
the
Hi Brian,
many thanks to you for your answers in the past! The always gave me the
little bit of mising information...
My Asterisk box is running fine now so I want to try the next step...
And now to all of you
What I want to implement is to use 1 button of my snom-360 phone for
following
I think its same as DND (do not disturb ) . It can be activated by *78 and deactivated by *79 . I use freepbx for configuration so i am not sure if its there in default asterisk setup . I snipped some part of my configuration from freepbx's config files
[app-dnd-on]exten = *78,1,Answerexten =
On 19:28, Sun 12 Nov 06, Vicky wrote:
I think its same as DND (do not disturb ) . It can be activated by *78 and
deactivated by *79 . I use freepbx for configuration so i am not sure if its
there in default asterisk setup . I snipped some part of my configuration
from freepbx's config files
Jamie Heckford wrote:
HOWEVER
It doesn't seem to send them to the *current* call. It places the
current call on hold and tries to place a call on a new line.
Currently looking for a workaround to this, will let you know.
Jamie,
Were you able to get a workaround on this? Just curious,
Hi,
The page can not be found!
Many thanks,
Christian
On 2006-11-12 at 05:39 Tzafrir Cohen wrote:
On Sat, Nov 11, 2006 at 09:42:27PM +0200, Tzafrir Cohen wrote:
I wonder, though, how that symlink was created. I hope an advice by me
was not involved...
I guess that more than just advice.
It seems like asterisk-addons in SVN has been broken for the last few
weeks:
gcc -DHAVE_CONFIG_H -I. -I. -I. -I./ooh323c/src -I./ooh323c/src/h323 -
DGNU -D_GNU_SOURCE -D_REENTRANT -D_COMPACT -c src/chan_h323.c -MT
chan_h323.lo -MD -MP -MF .deps/chan_h323.TPlo -fPIC -DPIC -o .libs/
I would like SayDigits to say a phone number faster. Is there a way
to control the speed?
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Hi,
I am looking for a TFTP server that is easy like the tftpd32 for Windows that I
have been using. Just want to start it with a command and my Cisco can connect
and retreive the config files from it.
Many thanks,
Christian
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-Original Message-
From: Christian [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Date: Sun, 12 Nov 2006 16:27:42 +0100
Subject: [asterisk-users] Looking for a simple TFTP server for Linux
Hi,
I am looking for a TFTP server that is easy like the tftpd32 for Windows
Most distros
I am planning to use asterisk with Digium TDM2404E card as a media
gateway to terminate traffic to Cell phones. Anyone got this working
before with no problmes, specially with Answer/Disconnect supervision?
Thanks
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Hello Doug,
I'm using Debian.
Many thanks,
Christian
On 2006-11-12 at 15:34 Doug Lytle wrote:
-Original Message-
From: Christian [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Date: Sun, 12 Nov 2006 16:27:42 +0100
Subject: [asterisk-users] Looking for a simple TFTP server for
On Sun, Nov 12, 2006 at 04:27:42PM +0100, Christian wrote:
Hi,
I am looking for a TFTP server that is easy like the tftpd32
for Windows that I have been using. Just want to start it with
a command and my Cisco can connect and retreive the config files from it.
Debian has:
tftpd
NZ == Norbert Zawodsky [EMAIL PROTECTED] writes:
NZ If I leave my desk I press this button. A light should show up as
NZ an indicator/reminder. From this moment all calls to my extension
NZ should immediately be transferred to my voicemail box.
NZ When I return I press the button again, the
BA == Benny Amorsen [EMAIL PROTECTED] writes:
BA An alternative is to make an extension which goes to voice mail
BA directly, and simply redirect the phone to that extension. It's a
BA bit more than one button, but at least the Snom 360 will show that
BA the redirection is active. Perhaps the
Some pictures from Astricon 2006 in Dallas.http://gallery.lith.za.net/Astricon-2006-- RegardsRob
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snip
I am running the most recent asterisk 1.2.13 on a Fedora 3.0.
When I go into asterisk (asterisk -r), defaults to verbose 3 and I get
a stream of messages:
Remote Unix connection
Remote Unix connection disconnected
...
...
-- end of file ---
has
Hey Rob, thanks for that. Brought back
good memories.
Cheers,
Dean
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rob Lith
Sent: Sunday, 12 November 2006
11:28 AM
To: Asterisk
Users Mailing List - Non-Commercial Discussion
Subject:
Hi,
perhaps someone has an answer for me:
- voicemail isn't sending any mails, even SMTP_SERVER and SMTP_Domain
are configured in rc.conf, and the mail address ist configured in
voicemail.conf.
- On my grandstream GPX2000 VM light flashes when there is a new
voicemail, but it doesn't go out
Most Linux distros have a TFTP server built in, however usually it functions through xinetd, which is probably already running on your machine, so actually it would cause no extra usage on your system unless the TFTP was in use. Check your distro's package repository and you should find something
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Martin Joseph
Sent: Saturday, November 11, 2006 15:01
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Re: Choppy sound in voicemail using
Asterisk1.2.11 on CENTOS4 guest
Hi everybody,
Well, I've finally got asterisk to to talk nicely with my Intertel pbx.
Currently there is a outside T1 line (em wink start, esf, b8zs)
connected to asterisk, and then asterisk connected similarly to my
Intertel pbx. For right now all asterisk is doing is passing calls
between
Does anyone know if there is a way to get a DB or
any other means to see if I can see if a call is coming from a cell phone or
not. If I am able to see if it is cellular or not is there any way to see aprox.
what area the phone is in (I know this wont be simple but would it work if I
have
I am having asterisk working with cdr mysql patch and freepbx for configurations . It stores all records in mysql tables and i can do further post paid billing myself . I am looking for a simple system that can show a user live call logs via web
What exactly are you trying to do? YOu can determine where the number was assigned and if it was originally a cell phone easily and cheapOn 11/12/06, Dovid B
[EMAIL PROTECTED] wrote:
Does anyone know if there is a way to get a DB or
any other means to see if I can see if a call is
With LNP in the US, there really is no way to determine if the call
originates from a landline, a VOIP line or a wireless line
Most numbers are portable and even the NPA doesn't tell much any more.
I have a wireless phone with a ( former ) landline number, and a VOIP
line with a ( former )
Sorry if you see this message repeated twice. I would like to set up
hard phones and softphones with the same extension so that when anybody
in the company dials an extension, each user's hardphone and softphone
will ring at the same time. I've tried setting this up before, but I
noticed that the
Matt,
as a start, what I can advise you is to take a tethereal trace and try
to reproduce the problem.
nohup tethereal host a.b.c.d -s2000 -w /tmp/yourtrace.cap
Where a.b.c.d is the IP address of your IP phone. You can then analyse
the trace and at least see if the asterisk box is sending
Check out the ESCAUX net.PBX operator console. In use in various
companies with 200+ extensions. Powerfull and convenient.
http://www.escaux.com//index.php?option=com_contenttask=viewid=61Itemid=350
Best Regards,
Jordi
--
www.escaux.com
Business IP Telephony
Forrest Beck wrote:
Talk to the
Depending on how you connect to the PSTN and what type of call is being
made, you may have access to the ANI II digits. The II digits tell you
what type user/service originated the call from such as: regular phone,
hotel/motel guest phone, pay phone, inmate phone, and various types of
It's for a call center. Calls are routed based on location. The customer
would rather the to be transferd without human interaction unless abolutely
nesc.
- Original Message -
From: John Novack [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
Am Sonntag, den 12.11.2006, 16:29 -0500 schrieb Jeronimo Romero:
Sorry if you see this message repeated twice. I would like to set up
hard phones and softphones with the same extension so that when anybody
in the company dials an extension, each user's hardphone and softphone
will ring at the
Just give ESCAUX net.PBX Free Edition a try.
You can start checking out our GUI at http://smp.free.escaux.com. These
web interfaces will generate the asterisk config files that are then
pushed to your asterisk box.
The full solution can be downloaded at http://www.escaux.com/netpbx
Have
Where can I get this info ?
- Original Message -
From:
Andrew
Joakimsen
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Sunday, November 12, 2006 11:18
PM
Subject: Re: [asterisk-users] Determine
if Call is from a cellular phone
What
Yes, the Mitel phones do have a Web interface for configuration.
They also support mass-deployment scenarios with TFTP HTTP.
You may want to check out these:
http://sipdnld.mitel.com/
http://edocs.mitel.com/DB/5212_5224/WebConfigHelp_Admin_en_CA/WebHelp/
WebConfig.htm
Thanks,
- Jesse
On
I edited zapata.conf to use custom ring cadences.
Seemed to work, but upon some restarts, seems zapata.conf is not being read
properly on startup
as zap show cadences will show the defaults. Some restarts show the custom
cadences.
What's up with that?
joe a.
Experiencing one way audio using IAX2.
I did see some other posts on this, and see there may be some internal issues
with asterisk and one way audio. Can this be a widespread problem? So many
seem to be using IAX, I find it puzzling.
Some information points to this being a problem on
Is this inherently an issue with sip? Is it possible for a sip server to
actually ring two different sip registration from the same account or is
this not possible under any sip enabled pbx?
Thanks again
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
This is not a SIP issue, but a problem with your configuration.
We have all hard phones register/authenticate with their MAC address as the
user/peer name. Soft phones use user id's that correspond to the person. We
then have our dialplan ring the appropriate devices (soft or hard) depending
on
Greetings all,
I'm playing with asterisk and two Polycom VSX300 videoconferencing
units. And I'm having zero luck getting video working over SIP.
The two units register fine with asterisk, and with allow=all in
sip.conf, the two units establish voice. But no video. And no obvious
messages
Oops,
Asterisk version is 1.2.12 (on Ubuntu)
On Mon, 2006-11-13 at 10:45 +1100, Peter Howard wrote:
Greetings all,
I'm playing with asterisk and two Polycom VSX300 videoconferencing
units. And I'm having zero luck getting video working over SIP.
The two units register fine with
I think puck.nether.net may still have a
txt file with the CO broken down by NPA-NXX. You can then look at the carrier
and know if it is Cell/LandLine.
You can also X-ref the CO-list and get
Lat/Long and or simply the zipcode to help you locate the caller. Not
perfect but unless the
On Mon, 2006-11-13 at 10:45 +1100, Peter Howard wrote:
Greetings all,
I'm playing with asterisk and two Polycom VSX300 videoconferencing
units. And I'm having zero luck getting video working over SIP.
The two units register fine with asterisk, and with allow=all in
sip.conf, the two
On Mon, 2006-11-13 at 00:57 +0100, Patrick wrote:
On Mon, 2006-11-13 at 10:45 +1100, Peter Howard wrote:
Greetings all,
I'm playing with asterisk and two Polycom VSX300 videoconferencing
units. And I'm having zero luck getting video working over SIP.
The two units register fine
I'm having difficulties getting zaptel to compile. I've compiled it in
the past and never had any difficulties to speak of, but on this
particular machine I have problems. The OS configuration is the same as
I've used in the past and the hardware is identical. Obviously there's
some subtle
Hi All...
My Asterisk system uses VoIP and also 2 POTS lines from Cox Communications.
Recently, the dial tone presentation from Cox seems to have slowed, so it
can take as long as 3 seconds to get a dial tone.
The problem I am having is that Asterisk does not seem to wait for the dial
tone
Jim Archer wrote:
Hi All...
My Asterisk system uses VoIP and also 2 POTS lines from Cox
Communications. Recently, the dial tone presentation from Cox seems to
have slowed, so it can take as long as 3 seconds to get a dial tone.
The problem I am having is that Asterisk does not seem to wait
--On Sunday, November 12, 2006 10:06 PM -0500 Steve Totaro
[EMAIL PROTECTED] wrote:
add a couple or few w's before you dial.
Okay, but where? I didn't see a w option for the dial command, and if I
add a wait before the dial won;t that just delay going off hook?
- Original Message -
From: Jim Archer [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, November 13, 2006 5:11 AM
Subject: Re: [asterisk-users] Slow to get dialtone when going off hook -big
problem for me :(
Hello All.
I am trying to use RAGI the ruby agi framework with trixbox. I am
having a problem
with the dialout part. The RAGI framework creates a file in the
/var/spool/asterisk/outgoing directory and routes the call to an
extension (I have listed the relevent portion of the file below). The
Jim Archer wrote:
Hi All...
My Asterisk system uses VoIP and also 2 POTS lines from Cox
Communications. Recently, the dial tone presentation from Cox seems to
have slowed, so it can take as long as 3 seconds to get a dial tone.
The problem I am having is that Asterisk does not seem to wait
- Original Message -
From: John Novack [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, November 13, 2006 6:46 AM
Subject: Re: [asterisk-users] Slow to get dialtone when going off hook -
bigproblem for me :(
Dovid B wrote:
snip
How hard would it be to have asterisk detect a dial tone ?
I really can't say. I am not a C programmer, so I wouldn't even know
where to start.
Given that cheap dial up modems have, for the past ??20?? years, have
been able to do just that, I would think it should have
Has anyone out there implemented a system that does call routing via Cisco
gear but VM for everyone on the system via Asterisk? What have been your
successes and failures or issues?
Thanks
Curt
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any logs/errors when you do a verbose 6 and a sip debug ?On 11/13/06, Peter Howard [EMAIL PROTECTED]
wrote:On Mon, 2006-11-13 at 00:57 +0100, Patrick wrote: On Mon, 2006-11-13 at 10:45 +1100, Peter Howard wrote:
Greetings all, I'm playing with asterisk and two Polycom VSX300 videoconferencing
--On Sunday, November 12, 2006 11:53 PM -0500 John Novack
[EMAIL PROTECTED] wrote:
Dovid B wrote:
snip
How hard would it be to have asterisk detect a dial tone ?
I really can't say. I am not a C programmer, so I wouldn't even know
where to start.
Given that cheap dial up modems have, for
Hello,
About my problem with CDR where 2 calls overlaps, and there is no evidence
that 3 other calls failed: after some searching on Asterisk bugs
database I have found:
http://bugs.digium.com/view.php?id=6762
...
When the Attended Transfer is used the information for call duration and
who is
On Mon, 2006-11-13 at 13:42 +0800, Rosli Sukri wrote:
any logs/errors when you do a verbose 6 and a sip debug ?
I've been running with verbose 9, debug 9, and sip debug. The resultant
output seems fine. The only warning is:
WARNING[6964]: chan_sip.c:3592 process_sdp: Unknown SDP media type
On Sun, Nov 12, 2006 at 07:26:02PM -0600, Traue, Paul wrote:
I'm having difficulties getting zaptel to compile. I've compiled it in
the past and never had any difficulties to speak of, but on this
particular machine I have problems. The OS configuration is the same as
I've used in the past
I'm trying to set up the Music on Hold feature.
However, when I place a call the moh starts and stops
immediately and as a result I dont hear the audio.
-- Started music on hold, class 'default', on
channel 'SIP/XXX'
-- Stopped music on hold on SIP/XXX
NOTICE[380]: res_musiconhold.c:515
Can two versions of asterisk run on same PC??
Keerthy,
Tr. Software Engineer,
PrimeSoft IP Solutions
Pvt. Ltd.,
Ph : 9246281937
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Can two versions of asterisk run on same PC??
Thanks in advance,
Keerthy
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