Pointing out a typo.. either in the mail or in the actual dialplan:-
exten = s,1,dial(local/2...@dialplan/n)
[dailplan]
--
Regards,
Prince Singh
W: http://www.drishti-soft.com
B: http://blog.drishti-soft.com
On Tue, Jun 30, 2009 at 8:27 PM, Benny Amorsen
2009/6/30 Carlos Chavez cur...@telecomabmex.com
On Tue, 2009-06-30 at 16:17 -0400, Jeremy Winder wrote:
I'm in the process of converting our current hybrid key system to
Asterisk and Aastra 57i phones. One of the features that seems to be a
show stopper for almost everyone in the office is
Hello, all. With the assistance of very helpful folks, our brand new
multi-tenant setup seems to be working smoothly from start to finish
with just a bump or two. The biggest is parking. Now that we got most
kinks worked out, I'm a little more comfortable in trying to resolve
this.
There seem
2 Things:-
1. Keep relevant subject line of the mails to public forums :)
2. Try direct IP call to another grandstream.
On Wed, Jul 1, 2009 at 5:56 AM, Todd Reese trees...@gmail.com wrote:
I did the upgrade to the phone. And the problem continued. Currently,
as per the previous
John F. Ervin wrote:
What do you do if you find things sharing interrupts (IRQ 11) in my
case with my X100P card. I believe there is some sort of internal
audio card in my cheap slow PC.
Check the BIOS whether you can:
Change the IRQ assignments
Disable the extra hardware using the same IRQ
On Wed, 2009-07-01 at 02:17 -0400, John A. Sullivan III wrote:
Hello, all. With the assistance of very helpful folks, our brand new
multi-tenant setup seems to be working smoothly from start to finish
with just a bump or two. The biggest is parking. Now that we got most
kinks worked out,
Here's how to configure this method properly
http://www.voip-info.org/wiki/view/Asterisk+callgroups+and+pickupgroups
Ish
Danny Nicholas wrote:
If it is configured and working correctly, *8 picks up the ringing line from
any eligible phone.
-Original Message-
From:
-- Forwarded message --
From: Xavier Cardil cardil.xav...@gmail.com
Date: Wed, Jul 1, 2009 at 10:51 AM
Subject: Unknown udp ports listening experts calling !
To: asterisk-users-requ...@lists.digium.com
Hello, last days we run under an very heavy issue with one audio stream
On Wed, Jul 1, 2009 at 7:37 AM, Francesco Peeters
france...@fampeeters.comwrote:
John F. Ervin wrote:
What do you do if you find things sharing interrupts (IRQ 11) in my
case with my X100P card. I believe there is some sort of internal
audio card in my cheap slow PC.
Check the BIOS
On 1 Jul 2009, at 09:54, Xavier Cardil wrote:
udp0 0 0.0.0.0:2727
0.0.0.0:* 4989/asterisk
udp0 0 0.0.0.0:9001
0.0.0.0:* 26354/udp-sender
udp0 0 0.0.0.0:5000
On Wed, Jul 1, 2009 at 5:09 AM, Tom O'Connor t...@twinhelix.org wrote:
On Wed, Jul 1, 2009 at 7:37 AM, Francesco Peeters
france...@fampeeters.com wrote:
John F. Ervin wrote:
What do you do if you find things sharing interrupts (IRQ 11) in my
case with my X100P card. I believe there is
On Wed, 2009-07-01 at 10:14 +0100, Steve Howes wrote:
On 1 Jul 2009, at 09:54, Xavier Cardil wrote:
udp0 0 0.0.0.0:2727
0.0.0.0:* 4989/asterisk
udp0 0 0.0.0.0:9001
0.0.0.0:*
Tom O'Connor wrote:
On Wed, Jul 1, 2009 at 7:37 AM, Francesco Peeters
france...@fampeeters.com mailto:france...@fampeeters.com wrote:
John F. Ervin wrote:
What do you do if you find things sharing interrupts (IRQ 11) in my
case with my X100P card. I believe there is some
On Wed, Jul 1, 2009 at 10:49 AM, Marco Signorini marcota...@libero.itwrote:
Tom O'Connor wrote:
On Wed, Jul 1, 2009 at 7:37 AM, Francesco Peeters
france...@fampeeters.com wrote:
John F. Ervin wrote:
What do you do if you find things sharing interrupts (IRQ 11) in my
case with my
I found nothing is passing through those ports . . . I think something was
sending the stream to our PST/SIP gateways, so the calls where affected when
getting in to the gateways. I found we are not running any extra TCL
applications on those gateways . . . could it be possible ? Could an UDP
On Wed, Jul 1, 2009 at 5:58 AM, Tom O'Connor t...@twinhelix.org wrote:
On Wed, Jul 1, 2009 at 10:49 AM, Marco Signorini marcota...@libero.itwrote:
Tom O'Connor wrote:
On Wed, Jul 1, 2009 at 7:37 AM, Francesco Peeters
france...@fampeeters.com wrote:
John F. Ervin wrote:
What do you
On Wed, Jul 1, 2009 at 6:08 AM, Steve Totaro stot...@first-notification.com
wrote:
On Wed, Jul 1, 2009 at 5:58 AM, Tom O'Connor t...@twinhelix.org wrote:
On Wed, Jul 1, 2009 at 10:49 AM, Marco Signorini marcota...@libero.itwrote:
Tom O'Connor wrote:
On Wed, Jul 1, 2009 at 7:37 AM,
On Wed, Jul 1, 2009 at 11:08 AM, Steve Totaro
stot...@first-notification.com wrote:
On Wed, Jul 1, 2009 at 5:58 AM, Tom O'Connor t...@twinhelix.org wrote:
On Wed, Jul 1, 2009 at 10:49 AM, Marco Signorini marcota...@libero.itwrote:
Tom O'Connor wrote:
On Wed, Jul 1, 2009 at 7:37 AM,
Xavier Cardil wrote:
I found nothing is passing through those ports . . . I think something
was sending the stream to our PST/SIP gateways, so the calls where
affected when getting in to the gateways. I found we are not running any
extra TCL applications on those gateways . . . could it be
You will need to insert the line before each place where you send calls
to Meetme and change the existing priority 1 to n. For example:
exten = 8600099,1,Playback(/var/lib/asterisk/sounds/silence/1)
exten = 8600099,n,Meetme(8600099)
exten =
Hi Bruce, thank you for your recommendations . . . I passed the test and the
only wanrning is this one :
/usr/sbin/unhide [ Warning ]
/usr/sbin/useradd[ OK ]
/usr/sbin/userdel
On Wed, Jul 1, 2009 at 1:10 AM, Olivieroza-4...@myamail.com wrote:
The 57i phone has 6 soft buttons which can show the status of at
least
16 phones (if you do not want to use the rest of the soft buttons which
would give you another 16).
Are you sure of that ?
How can you set more
On Tue, Jun 30, 2009 at 11:53 PM, John A. Sullivan III
jsulli...@opensourcedevel.com wrote:
On Wed, 2009-07-01 at 02:17 -0400, John A. Sullivan III wrote:
Hello, all. With the assistance of very helpful folks, our brand new
multi-tenant setup seems to be working smoothly from start to
This has been fixed in the 1.6.1 SVN, and you will have to back
port a patch until these changes are rolled into another release. I was
disappointed that more bug fixes were not included in 1.6.1.1.
-Jonathan
Asterisk 1.6.1.1 was released
On 29/06/09 18:26, Gordon Henderson wrote:
Looking for a (windows) app. that will listen to the manager interface
then pop-up a web browser pointing to a page on an incoming phone call..
Not looking for outlook integration, or outbound dialling, just to
recognise an incoming call and poke a
On Wed, 2009-07-01 at 07:17 -0700, Jonathan Thurman wrote:
On Tue, Jun 30, 2009 at 11:53 PM, John A. Sullivan III
jsulli...@opensourcedevel.com wrote:
On Wed, 2009-07-01 at 02:17 -0400, John A. Sullivan III wrote:
Hello, all. With the assistance of very helpful
Any more suggestions ?
On Wed, Jul 1, 2009 at 8:30 AM, David @ULC ucoms2...@gmail.com wrote:
Thanks for the Reply,
I was waiting online for someone to reply : -)
Here is my Extension file : [ Where should I enter those line ? ]
exten = 8600099,1,Meetme(8600099)
exten =
What was the result of my earlier suggestion? See below.
Joshua Billings wrote:
You will need to insert the line before each place where you send
calls to Meetme and change the existing priority 1 to n. For example:
exten = 8600099,1,Playback(/var/lib/asterisk/sounds/silence/1)
exten =
Hi, all. I've got an old Telrad PBX with an Emagen(?) voicemail box. The
VM box, itself, is beginning to show its age. Big-time. We're thinking it
might be time to look for a replacement. I'd love to install Asterisk
with an FXO card or something, but I don't think it supports whatever
Have you verified that the sound file is intact (convert to wav with sox and
play thru mplayer, or just set up a test line exten =
7529,1,Playback(conf-onlyperson)?
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David @ULC
On Wed, 1 Jul 2009, Alan Lord (News) wrote:
On 29/06/09 18:26, Gordon Henderson wrote:
Looking for a (windows) app. that will listen to the manager interface
then pop-up a web browser pointing to a page on an incoming phone call..
Not looking for outlook integration, or outbound dialling,
This has been fixed in the 1.6.1 SVN, and you will have to back port a
patch until these changes are rolled into another release. I was
disappointed that more bug fixes were not included in 1.6.1.1.
-Jonathan
Asterisk 1.6.1.1 was released for a security issue, AST-2009-001. Why
would
On Wed, Jul 1, 2009 at 11:16 AM, Ken D'Ambrosio k...@jots.org wrote:
Hi, all. I've got an old Telrad PBX with an Emagen(?) voicemail box. The
VM box, itself, is beginning to show its age. Big-time. We're thinking it
might be time to look for a replacement. I'd love to install Asterisk
*
*
*/var/lib/asterisk/sounds/silence/1*
*
*
*1 is the folder or the filename ?*
On Wed, Jul 1, 2009 at 8:30 AM, David @ULC ucoms2...@gmail.com wrote:
Thanks for the Reply,
I was waiting online for someone to reply : -)
Here is my Extension file : [ Where should I enter those line ? ]
sound file is intact
Yes. I checked it with my other server.
On Wed, Jul 1, 2009 at 9:14 PM, David @ULC ucoms2...@gmail.com wrote:
*
*
*/var/lib/asterisk/sounds/silence/1*
*
*
*1 is the folder or the filename ?*
On Wed, Jul 1, 2009 at 8:30 AM, David @ULC ucoms2...@gmail.com wrote:
1 is the filename. The Playback application does not require you to
specify the extension. The idea is that by playing 1 second of silence
the message for MeetMe remains intact. Let me know how it goes.
- Josh
David @ULC wrote:
/
/
//var/lib/asterisk/sounds/silence/1/
/
/
/1 is the
Could someone tell me how to set which IRQ the ISDN card picks up?
It's a multi-stage process.
Each PCI slot has four interrupt pins: INTA through INTD. A
PCI card can choose to use any of these four (or even more than
one of them, as some multi-port serial cards do). Most PCI cards
use only
*This is what I get when I reload in CLI :*
== Parsing '/etc/asterisk/extconfig.conf': Found
== Parsing '/etc/asterisk/manager.conf': Found
== Parsing '/etc/asterisk/cdr.conf': Found
Jul 1 12:05:59 NOTICE[23347]: cdr.c:1214 do_reload: CDR simple logging
enabled.
== Parsing
On Wed, 1 Jul 2009, Ken D'Ambrosio wrote:
Hi, all. I've got an old Telrad PBX with an Emagen(?) voicemail box. The
VM box, itself, is beginning to show its age. Big-time. We're thinking it
might be time to look for a replacement. I'd love to install Asterisk
with an FXO card or
On 01/07/09 16:29, Gordon Henderson wrote:
On Wed, 1 Jul 2009, Alan Lord (News) wrote:
On 29/06/09 18:26, Gordon Henderson wrote:
Looking for a (windows) app. that will listen to the manager interface
then pop-up a web browser pointing to a page on an incoming phone call..
Not looking for
Dave Platt wrote:
Could someone tell me how to set which IRQ the ISDN card picks up?
It's a multi-stage process.
Each PCI slot has four interrupt pins: INTA through INTD. A
PCI card can choose to use any of these four (or even more than
..
bridge architecture might be
As others have said, this is certainly possible. Our old NEC phone
system had us in the same boat. It triggered voicemail by ringing
the VM extension(s) and sending a DTMF burst of the extension to
record VM for within 1.5 seconds. In our case, when any call came it
in went to the
Make a call to VM (has to go out on the port you have the handset plugged
into), answer it and listen.
If you hear a bunch of DTMF then you are golden.
Sounds like good stuff, but my most substantial concerns involved things
like MWI: is asterisk able to push that back to the PBX?
--
Hi All;
How can I test manager.conf?
Can I telnet to the asterisk machine at the port 5038 and send and receive
commands to test if the manager is working fine? How?
Regards
Bilal
___
-- Bandwidth and Colocation Provided by
On Wed, Jul 1, 2009 at 12:15 PM, Jeff LaCoursiere j...@jeff.net wrote:
On Wed, 1 Jul 2009, Ken D'Ambrosio wrote:
Hi, all. I've got an old Telrad PBX with an Emagen(?) voicemail box.
The
VM box, itself, is beginning to show its age. Big-time. We're thinking
it
might be time to look
- Marco Signorini marcota...@libero.it wrote:
Dave Platt wrote:
Could someone tell me how to set which IRQ the ISDN card picks up? It's a
multi-stage process.
Each PCI slot has four interrupt pins: INTA through INTD. A
PCI card can choose to use any of these four (or even more than
On Wed, Jul 1, 2009 at 6:35 AM, Tom O'Connor t...@twinhelix.org wrote:
On Wed, Jul 1, 2009 at 11:08 AM, Steve Totaro
stot...@first-notification.com wrote:
On Wed, Jul 1, 2009 at 5:58 AM, Tom O'Connor t...@twinhelix.org wrote:
On Wed, Jul 1, 2009 at 10:49 AM, Marco Signorini
For those of you who have been waiting for ATT to announce the public
availability of their femtocell appliance in order to fix the shitty ATT
network coverage this will interest you.
Vodafone Access Gateway (femtocell) launched in UK
http://www.abiresearch.com/Blog/Wireless_Blog/635
On Wed, 1 Jul 2009, bilal ghayyad wrote:
Hi All;
How can I test manager.conf?
Can I telnet to the asterisk machine at the port 5038 and send and
receive commands to test if the manager is working fine? How?
Yes!
RTFM would be a fine place to start - or at least the wiki:
On Wed, 1 Jul 2009, Dean Collins wrote:
For those of you who have been waiting for ATT to announce the public
availability of their femtocell appliance in order to fix the shitty
ATT network coverage this will interest you.
It's getting a lot of press and a bit of a mixed reaction over
agreed.
extended o2 coverage would be very useful, especially for Wales!
I like the idea, i don't like the idea of paying, if they want mobile
traffic it should be possible to buy your own hardware controlled in the
same method as wireless AP's allowing you to connect for free to the service
and
2009/7/1 Geraint Lee gera...@gmail.com:
agreed.
extended o2 coverage would be very useful, especially for Wales!
I like the idea, i don't like the idea of paying, if they want mobile
traffic it should be possible to buy your own hardware controlled in the
same method as wireless AP's
Hello List,
I'm having problems with registrating my Asterisk-server to the
SIP-provider. Yesterday all worked fine, this evening I cannot call out.
What can be wrong ?
This is my registration in sip.conf :
register = 092779077:x...@85.119.188.3
This the output of SIP show peers :
On Wed, 2009-07-01 at 13:05 -0400, Ken D'Ambrosio wrote:
Sounds like good stuff, but my most substantial concerns involved things
like MWI: is asterisk able to push that back to the PBX?
Does your existing PBX use SMDI to interface with your current voicemail
system? If so, recent versions of
On Wed, 2009-07-01 at 10:25 -0700, bilal ghayyad wrote:
Can I telnet to the asterisk machine at the port 5038 and send and receive
commands to test if the manager is working fine?
Absolutely!
How?
1) Make sure manager is enabled in manager.conf (enabled=yes in
[general] section)
2) Create
jonas kellens wrote:
Hello List,
I'm having problems with registrating my Asterisk-server to the
SIP-provider. Yesterday all worked fine, this evening I cannot call out.
What can be wrong ?
This is my registration in sip.conf :
register = 092779077:x...@85.119.188.3
Reliably Transmitting (no NAT)
and you are natted I presume (
Port 5060 is forwarded to the internal IP-address of my
Asterisk-server).
Another Belgian user :)
Olivier
jonas kellens a écrit :
Hello List,
I'm having problems with registrating my Asterisk-server to the
Hello!
I have a sip device that is sending in the SDP:
rtpmap:98 g729a
It does not seem like Asterisk is negotiating the codec properly,
because while the call rings, the rtp lines fail. However, on other
sip devices that have rtpmap:18 g729 in their SDP, things work fine
with Digium's
Wow. Thanks for all the replies! Something just occurred to me, though:
which side would be FXO, and which side would be FXS? The PBX? Or the
Asterisk/VM side?
Thanks again for all the info!
-Ken
On Wed, July 1, 2009 3:36 pm, Jared Smith wrote:
On Wed, 2009-07-01 at 13:05 -0400, Ken
Elliot Murdock wrote:
Hello!
I have a sip device that is sending in the SDP:
rtpmap:98 g729a
It does not seem like Asterisk is negotiating the codec properly,
because while the call rings, the rtp lines fail. However, on other
sip devices that have rtpmap:18 g729 in their SDP, things
The PBX would be FXS since it originates the calls, * would be FXO since it
only receives calls in this case.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ken D'Ambrosio
Sent: Wednesday, July 01, 2009 4:06
2009/7/1 Ken D'Ambrosio k...@jots.org
Wow. Thanks for all the replies! Something just occurred to me, though:
which side would be FXO, and which side would be FXS? The PBX? Or the
Asterisk/VM side?
It seems PBX should be equiped with FXO interface(s) and Asterisk with FXS
ones.
2009/7/1 Jonathan Moore supermegat...@gmail.com
On Wed, Jul 1, 2009 at 1:10 AM, Olivieroza-4...@myamail.com wrote:
The 57i phone has 6 soft buttons which can show the status of at
least
16 phones (if you do not want to use the rest of the soft buttons which
would give you another
2009/7/1 Danny Nicholas da...@debsinc.com
The PBX would be FXS since it originates the calls, * would be FXO since it
only receives calls in this case.
Yes you're right : if Asterisk behaves like a phone, it should plus into
PBX's FXS ports (and so be equiped with FXO ports).
Sorry, for my
Looks like you might be getting conflicting information.
The important thing is the * ports must be opposite what the PBX ports are.
Odds are, the ports on the PBX used for voice-mail are extension ports--they
look like central office lines (PSTN POTS), providing dial tone. The ports
on the
On Wed, Jul 1, 2009 at 3:36 PM, Jared Smithjsm...@digium.com wrote:
On Wed, 2009-07-01 at 13:05 -0400, Ken D'Ambrosio wrote:
Sounds like good stuff, but my most substantial concerns involved things
like MWI: is asterisk able to push that back to the PBX?
Does your existing PBX use SMDI to
Hi All;
I download asterisk, compiled it and install it, but not finding the sounds
file (/var/lib/asterisk/sounds), what could be the reason and how I can have it
without repeating every thing?
My asterisk version is: Asterisk 1.4.25
Regards
Bilal
On Tue, Jun 30, 2009 at 10:47 AM, Ade
Vickersaster...@solutionengineers.com wrote:
I run a phone in a remote office using the IAX2 protocol. It mostly works
fine; except that every 5 mins it loses connection with Asterisk, before
reconnecting 30 seconds later; rinse repeat.
I used to have
On Tue, Jun 30, 2009 at 9:21 AM, Deric Pagederic.p...@nisc.coop wrote:
I've set up an outbound .call system for customer callbacks and the like.
Calls are going out over analog lines and I'm trying to use the
WaitForSilence routine to make sure the phone has stopped ringing before
starting
Check out http://www.voip-info.org/wiki/view/Asterisk+iax+qualify.
I've ran into problems with home routers not keeping the connection alive, udp
timeouts most likely. These options particularly, the qualifyfreqnotok will
have asterisk send out a poke to the soft phone if it reports the phone
On Wed, Jul 1, 2009 at 4:40 PM, Olivieroza-4...@myamail.com wrote:
True but how can a single light be blinking because extension 1001 is
receiving a call and at the same time, be turned on because extension 1002
is on call ?
Maybe typing on Next button would alternatively show extension 1001
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