Hi all.
Got a situation here. I'm trying to setup IPPBX at few locations and I
need to peer those locations so that the users from one location can call
the users of other locations over IP network. have setup Grandstream
GXE5024 and one end and a Trixbox 2.6.0.7 at another end. problem is I
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Jerome Poggi wrote:
Yesturday I found a bug in Asterisk, in particular in Dial application.
When the Dial function exit it want to branch to n+1, but if n+1 do not
exist, it exit from the context.
Example :
exten = s,5,ChanIsAvail(SIP/604,s)
Hi Matt!!
Thanks for that. When i use the same config, it looks like my Asterisk
1.4.21.1 really expects the md5secret because i get this :
[Jul 7 11:11:21] NOTICE[17678]: chan_sip.c:15236 handle_request_register:
Registration from 'sip:[EMAIL PROTECTED]:5060' failed for
'10.10.250.252' -
Hi all,
i'm trouble with codec setup on an asterisk machine 1.4.18 and some
Thomson ST2030 as extensions.
In the users.conf file for internal extension i have:
disallow=all
allow=g729
allow=alaw
allow=ulaw
Without any codec installed (i mean with original g729 of asterisk)
all go fine, calling
You didn't give details of your networking setup but do you have the
3102 and then X-Lite client connected to the same switch or router? It
not, one switch could be dropping packets or slow. Do you qualify both
devices in Asterisk? Do they have the same ping times?
I haven't done any audio
Hi,
I'm currently using GXW-4008 from Grandstream.
I would like to know if anyone can recommend another 8-FXS-port with 2 RJ-45
ethernet ports ATA gateway.
I would like to stress on the 2 RJ-45 ethernet WAN/LAN ports as they allow me
to fail over another switch in case the first malfunctions
If my memory serves me right, we could use Thomson ST2030 and Asterisk 1.4.
Have you tried with another soft or hardphone ?
2008/7/7 Vinz486 [EMAIL PROTECTED]:
Hi all,
i'm trouble with codec setup on an asterisk machine 1.4.18 and some
Thomson ST2030 as extensions.
In the users.conf file
I have a realtime queue and the state of the queue member change as
below. Not-in-use (no call)- Unknown (ringing)- Not-in-use
(answered). The state shown in show queues does not really reflect
the state of the phone. I have searched the net and also the
UPGRADE.TXT by the warning message
Hi folks,
we use meetme application with pin so when a customer joins he's
prompted for his name.
Then the voice say:press one to accept the recording...
My question is, is it possible to cut off that request topress one?
Thanks to all
--
.:FaberK:.
On Mon, Jul 7, 2008 at 6:58 AM, FaberK [EMAIL PROTECTED] wrote:
Hi folks,
we use meetme application with pin so when a customer joins he's
prompted for his name.
Then the voice say:press one to accept the recording...
My question is, is it possible to cut off that request topress one?
Steve Totaro wrote:
On Mon, Jul 7, 2008 at 6:58 AM, FaberK [EMAIL PROTECTED] wrote:
Hi folks,
we use meetme application with pin so when a customer joins he's
prompted for his name.
Then the voice say:press one to accept the recording...
My question is, is it possible to cut off that request
FaberK schrieb:
we use meetme application with pin so when a customer joins he's
prompted for his name.
Then the voice say:press one to accept the recording...
My question is, is it possible to cut off that request topress one?
Audacity. Edit the sound file.
Grüße,
Philipp Kempgen
--
Hello!
We would like to receive a SIP call and keep the caller waiting
listening to some music other sound. A secondary intelligence decides
whom to connect to and creates an outbound SIP call and when it is
ringing there, or after the recipient answered the call, and maybe after
listening to
Hi,
but if I edit the sound file, remain that I have to press the 1
button to go ahead.
Thanks to all.
On Mon, Jul 7, 2008 at 1:57 PM, Philipp Kempgen
[EMAIL PROTECTED] wrote:
FaberK schrieb:
we use meetme application with pin so when a customer joins he's
prompted for his name.
Then the
On Mon, Jul 7, 2008 at 12:18 PM, Olivier [EMAIL PROTECTED] wrote:
If my memory serves me right, we could use Thomson ST2030 and Asterisk 1.4.
Have you tried with another soft or hardphone ?
Why not???
___
-- Bandwidth and Colocation Provided by
FaberK schrieb:
but if I edit the sound file, remain that I have to press the 1
button to go ahead.
Thanks to all.
On Mon, Jul 7, 2008 at 1:57 PM, Philipp Kempgen
[EMAIL PROTECTED] wrote:
FaberK schrieb:
we use meetme application with pin so when a customer joins he's
prompted for
Hi!
FaberK schrieb:
My question is, is it possible to cut off that request topress one?
I think you want to get rid of the number-pressing. The only option to
omit this seems to be option E - select an empty pinless conference.
http://www.voip-info.org/wiki-Asterisk+cmd+MeetMe
Regards
Hi!
I'm using asterisk 1.4.17 with twinkle and a custom phone based on
iaxclient 2.0.2 and I'm struggling a bit with DTMF and features.conf.
While the twinkle client is able to initiate an attended transfer using
*2 (as configured in features.conf), the iax client is not. I can see
the DTMF
- Eric \ManxPower\ Wieling [EMAIL PROTECTED] wrote:
Tell your box to not expect Caller*ID information. You set that with
usercallerid=no in /etc/asterisk/zapata.conf
Since you are using the Asterisk Appliance you would have to contact
Digium for support.
Sydney Web Hosting wrote:
Your best option is to use queues. If for some raison you can't use queues
you'll need to do some serious programming (agi, manager api) to get things
working. You can probably do the basic stuff using dialplan logic and a few
shell scripts, but you'll need to get a lot more involved when
have you tried app_queue?
On 7/7/08, Philipp Ott [EMAIL PROTECTED] wrote:
Hello!
We would like to receive a SIP call and keep the caller waiting
listening to some music other sound. A secondary intelligence decides
whom to connect to and creates an outbound SIP call and when it is
ringing
On Mon, Jul 7, 2008 at 2:48 PM, Philipp Ott [EMAIL PROTECTED] wrote:
Hi!
FaberK schrieb:
My question is, is it possible to cut off that request topress one?
I think you want to get rid of the number-pressing. The only option to
omit this seems to be option E - select an empty pinless
Hi Kashif,
DID World Wide (www.didww.com) would be happy to provide DIDs at $5.00/month
for Paris (prefix 1) and $8.00/month for Gottenburg (prefix 31). Setup is $5.00
per DID. Calling is unlimited inbound. Note that this is low-volume pricing,
and discounts would apply on purchases above 10
Hi, I want to use any java open source solution to implement click-to talk
in my web page connected to my Asterisk.
I don´t need a callback solution.
Regards.
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 -
Rilawich Ango wrote:
I have a realtime queue and the state of the queue member change as
below. Not-in-use (no call)- Unknown (ringing)- Not-in-use
(answered). The state shown in show queues does not really reflect
the state of the phone. I have searched the net and also the
UPGRADE.TXT by
Is it possible for music on hold to be in a central server? upgrading to
1.6 is not an option for me currently.
i have 2 asterisk servers, i'm using DNS SRV, i have a web interface
where user can upload their own music on hold, but forgot that when they
upload it wont be uploaded on the
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Nhadie wrote:
| Is it possible for music on hold to be in a central server? upgrading to
| 1.6 is not an option for me currently.
|
If the MOH is on server 1, you could export that MOH directory to server
2 via NFS.
Barry
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I've been having a problem with Asterisk MWI notification on my SIP
phones since going to version 1.4 a long time ago. Since going to this
version, I have needed to go into chan_sip.c and do the following:
/*! \brief Check whether peer needs a new MWI notification check */
static int
I'm using i6net's vxml browser in Asterisk.
I'm trying to work
out how I can return the inputs from a menu or form back into the
Asterisk dial plan. Is there a variable?
The exit tag apparently can be used to return a value (still trying to
work out how to do that), but what about multiple
Hi,
I need help with modifying cdr_addon_mysql.c I want to have more
fields in cdr table in asterisk. I've tried to modify cdr_addon_mysql.c
and replace userfield with ex team (sed -e 's/userfield/team/g' ). When
I try to recomplie
menuselect/menuselect --check-deps
Anybody have ideas on how I can troubleshoot? From what I've read
cisco VoIP phones should be able to get PoE from these switches. I'm
using a straight-through cat5e cable. Plug the phone in and nothing.
Is there anyway I can test the PoE switch (it was a refurb unit from
CDW) w/out having a
So, I need to build a complicated IVR with Asterisk, with a lot of back end
hooks. The dial plan itself has a lot of limitations, not the least of which is
that the dial plan is ugly, hard to maintain, and full of gotchas like all
variables being global etc etc.
I've been involved with
On 7/7/08, M B [EMAIL PROTECTED] wrote:
Anybody have ideas on how I can troubleshoot? From what I've read
cisco VoIP phones should be able to get PoE from these switches. I'm
using a straight-through cat5e cable. Plug the phone in and nothing.
Is there anyway I can test the PoE switch
Hi Roland,
I think there is an issue with the screen refresh, mine also displays
searching... unless I reboot the phone, and leave wifi on when it boots
up, at this point it says internet calling: available .. but, it works
either way.
As for prepending a 9, that's something your Asterisk
If it is an older pre-802.3af phone it wants CDP, you will need to
crimp a reverse polarity cable or buy an online CDP converter. Search
cisco cable on voip-info
Cory Andrews
Director of New Business Initiatives
-
Sayers Media Group
[EMAIL PROTECTED]
On Mon, Jul 7, 2008 at 1:21 PM, Douglas Garstang [EMAIL PROTECTED] wrote:
So, I need to build a complicated IVR with Asterisk, with a lot of back end
hooks. The dial plan itself has a lot of limitations, not the least of which
is that the dial plan is ugly, hard to maintain, and full of gotchas
On Monday 07 July 2008 12:05:16 Marcin J. Kowalczyk wrote:
I need help with modifying cdr_addon_mysql.c I want to have more
fields in cdr table in asterisk.
snip
Any idea how to solve this problem?
Use the one in trunk. It already supports arbitrary fields, no source change
necessary.
Hmm...
You may be in one of those positions where there just isn't a great
solution because your environment has so many constraints. You might
want to check out the way freeswitch handles IVRs, dialplan hooks,
FAGI-ish connections, etc. It will still take some work, of course,
because there
Yes, everything is connected to the same switch - which is a very high
performance Cisco device. Ping times are the same, well under 1 ms, no dropped
packets. Network appears clean. Here is the sip configuration:
[sipura1_line1]
type=friend
username=sipura1_line1
secret=xxx
host=dynamic
I am using asterisk 1.4.21and svn-124910 and getting the chan_alsa:693
resource temporarily unavailable message.
The audio is working but I dont recall getting any error message in the
past.
Is this something to be concerned about?
Jerry
___
--
Tilghman Lesher pisze:
Use the one in trunk. It already supports arbitrary fields, no source change
necessary. Motivate the right person, and it may even get backported to 1.4
(or backport it yourself).
I've tried to use:
svn co http://svn.digium.com/svn/asterisk-addons/trunk/
This is some pretty basic stuff... (someone will probably send you a RTFM)
Start with the sample dialplan (make samples I think)...trace the dialplan
along to understand how it works
Check the wiki and then post anything that you need help with
From: [EMAIL PROTECTED]
[mailto:[EMAIL
On Monday 07 July 2008 14:22:25 Marcin J. Kowalczyk wrote:
Tilghman Lesher pisze:
Use the one in trunk. It already supports arbitrary fields, no source
change necessary. Motivate the right person, and it may even get
backported to 1.4 (or backport it yourself).
I've tried to use:
We are in the process of preparing to move our Asterisk server to a
Digital T1 interface card instead of a analog card (via an Adtran
which is now connected to the T1). I did a preliminary test the other
day and hooked the T1 line up to the T1 card, bypassing the Adtran.
This worked
Digital ISDN used Q931 messages. You should get a disconnect message
from telco on the d-channel 23.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Daniel
Hazelbaker
Sent: Monday, July 07, 2008 4:39 PM
To: Asterisk Users Mailing List - Non-Commercial
Daniel Hazelbaker wrote:
We are in the process of preparing to move our Asterisk server to a
Digital T1 interface card instead of a analog card (via an Adtran
which is now connected to the T1). I did a preliminary test the other
A T1 or a PRI? Just make sure we're on the same page.
Perhaps you could try the OpenSUSE LiveCD and find out.
-Mark Best
-Network Administrator
[EMAIL PROTECTED]
-(208) 750-2054
This communication is the property of Nez Perce County and may contain
confidential or privileged information. The information contained in
this
I have the option of running either SIP or SCCP for my cisco VoIP
rollout..can someone shed light on what the pros/cons are? Seems
everything is SIP these days so that's the option im leaning. Thanks-
Matt
___
-- Bandwidth and Colocation Provided by
I'm trying to get some useful status info on Asterisk queues. Using the
Asterisk::Manager perl module, I've attempted to gather Queues and
QueueStatus, but neither are useful. In fact, Queues only returns one
out of four possible queues.
I found references online to QueueMemberStatus, which is
Good evening all - for the first time, I'm implementing my first-ever
queue in asterisk. Overall, it's a pretty simple setup, 4 static
members, very low call volume, etc. The one thing that has stumped me
so far, though, is the following...
This is a queue I'm setting up for contacting our IT
Jason Dixon wrote:
I'm trying to get some useful status info on Asterisk queues. Using the
Asterisk::Manager perl module, I've attempted to gather Queues and
QueueStatus, but neither are useful. In fact, Queues only returns one
out of four possible queues.
I found references online to
Jason Dixon schrieb:
I'm trying to get some useful status info on Asterisk queues. Using the
Asterisk::Manager perl module, I've attempted to gather Queues and
QueueStatus, but neither are useful. In fact, Queues only returns one
out of four possible queues.
I found references online to
On Mon, Jul 7, 2008 at 5:31 PM, M B [EMAIL PROTECTED] wrote:
I have the option of running either SIP or SCCP for my cisco VoIP
rollout..can someone shed light on what the pros/cons are? Seems
everything is SIP these days so that's the option im leaning. Thanks-
I'm not sure how this question
Hi,
I'm trying to setup Asterisk as an outgoing SIP dial tester. There will
be no phones connected to this installation, and I don't need to
process incoming calls. I just need to dial a number, have the person
acknowledge the call, and log that fact. (Basically an automated soft
phone). I
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Vieri wrote:
Hi,
I'm currently using GXW-4008 from Grandstream.
I would like to know if anyone can recommend another 8-FXS-port with 2 RJ-45
ethernet ports ATA gateway.
I would like to stress on the 2 RJ-45 ethernet WAN/LAN ports as they
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FaberK wrote:
On Mon, Jul 7, 2008 at 2:48 PM, Philipp Ott [EMAIL PROTECTED] wrote:
Hi!
FaberK schrieb:
My question is, is it possible to cut off that request topress one?
I think you want to get rid of the number-pressing. The only option to
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Florian Hackenberger wrote:
Hi!
I'm using asterisk 1.4.17 with twinkle and a custom phone based on
iaxclient 2.0.2 and I'm struggling a bit with DTMF and features.conf.
Maybe the feature digit timeout?
- --
Kind Regards,
Matt Riddell
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Joseph Jacobson wrote:
Hi,
I'm trying to setup Asterisk as an outgoing SIP dial tester. There will
be no phones connected to this installation, and I don't need to
process incoming calls. I just need to dial a number, have the person
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Erik Anderson wrote:
Good evening all - for the first time, I'm implementing my first-ever
queue in asterisk. Overall, it's a pretty simple setup, 4 static
members, very low call volume, etc. The one thing that has stumped me
so far, though, is
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Tilghman Lesher wrote:
On Thursday 03 July 2008 00:27:00 Matt Riddell wrote:
Tilghman Lesher wrote:
I find that a good number of people are using . in a pattern in
situations that are entirely unnecessary (such as local numbers). The
only place
Hi. I'm writing a speech recognition engine for Asterisk. I'm having a
problem with ast_speech_write, the audio data is coming with a silence of
0.20ms at each between 0.20ms of normal audio. The audio data configured in
ast_speech_new is AST_FORMAT_SLINEAR. Am I wasting some needed
configuration?
On 07/08/08 11:05, Matt Riddell wrote:
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Joseph Jacobson wrote:
Hi,
I'm trying to setup Asterisk as an outgoing SIP dial tester. There will
be no phones connected to this installation, and I don't need to
process incoming calls. I just need to
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Ok, if you type:
core set verbose 10
and
core set debug 10
Then drop the file into /var/spool/asterisk/outgoing
a) does the file disappear
b) does anything come up in the console
c) what is the date on the file i.e.:
send us
ls -alh
On Tue, Jul 08, 2008 at 12:32:17AM +0200, Philipp Kempgen wrote:
Jason Dixon schrieb:
I'm trying to get some useful status info on Asterisk queues. Using the
Asterisk::Manager perl module, I've attempted to gather Queues and
QueueStatus, but neither are useful. In fact, Queues only
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Jason Dixon wrote:
On Tue, Jul 08, 2008 at 12:32:17AM +0200, Philipp Kempgen wrote:
Jason Dixon schrieb:
I'm trying to get some useful status info on Asterisk queues. Using the
Asterisk::Manager perl module, I've attempted to gather Queues and
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Hi,
I've completed the first stage of the AsteriskWatch FaceBook application:
http://apps.facebook.com/asterisk/
It encompasses a wide range of Asterisk features including Karma, News,
Discussion Boards and Links.
It will add a box to your profile
Hi All,
I was under the impression that I found a WEB site about two years or so
ago that allowed Asterisk users to place free calls between each other
that used up users un-used minutes/calls. I though the site was IAXtel
but that does not seem to be the case.
As an example I have a plan
- Original Message -
From: spectro [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, July 01, 2008 8:02 PM
Subject: Re: [asterisk-users] sip extension compromised,need help blocking
brute force attempts
On
On 07/08/08 11:55, Matt Riddell wrote:
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Ok, if you type:
core set verbose 10
and
core set debug 10
Then drop the file into /var/spool/asterisk/outgoing
a) does the file disappear
b) does anything come up in the console
c) what is the date on the
PROTECTED]:3] Set(SIP/1XX-007321b0,
TIMEOUT(absolute)=3600) in new stack
-- Channel will hangup at 2008-07-08 02:53:51 UTC.
-- Executing [EMAIL PROTECTED]:4] Set(SIP/1XX-007321b0,
FAXFILE=/var/spool/asterisk-fax/20080707-195351-1xx.tif) in
new stack
Klaverstyn, David C wrote:
Hi All,
I was under the impression that I found a WEB site about two years or so
ago that allowed Asterisk users to place free calls between each other
that used up users un-used minutes/calls. I though the site was IAXtel
but that does not seem to be the
-- Executing [EMAIL PROTECTED]:3] Set(SIP/1XX-007321b0,
TIMEOUT(absolute)=3600) in new stack
-- Channel will hangup at 2008-07-08 02:53:51 UTC.
-- Executing [EMAIL PROTECTED]:4] Set(SIP/1XX-007321b0,
FAXFILE=/var/spool/asterisk-fax/20080707-195351-1xx.tif) in
new stack
That was Free World Dialup's service called Bellster which then
changed names to fwdOUT. Yes, it was against the EULA of various
VOIP providers.
You can still find the ghost ship pages here:
http://www.fwdout.com/web/ - can someone find out if this still
works or not? It was an
On 07/07/08 21:40, Joseph Jacobson wrote:
On 07/08/08 11:55, Matt Riddell wrote:
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Ok, if you type:
core set verbose 10
and
core set debug 10
Then drop the file into /var/spool/asterisk/outgoing
a) does the file disappear
b) does anything come up
Hi,
I'd like to know if someone already succesfully installed sippyskype.
Here on gentoo, I'm starting to build necessary stuff.
1) mjsip_1.6
I used :
http://blogimg.chinaunix.net/blog/upfile/070801012614.pdf
[EMAIL PROTECTED] ~/Documents/Perso/Voip/SippySkype/mjsip_1.6 $ make all
make sip
On Mon, Jul 7, 2008 at 11:28 PM, Steve Totaro
[EMAIL PROTECTED] wrote:
On Mon, Jul 7, 2008 at 10:46 AM, Emmanuel Favre-Nicolin
[EMAIL PROTECTED] wrote:
Hi,
I'd like to know if someone already succesfully installed sippyskype.
Here on gentoo, I'm starting to build necessary stuff.
1)
On Mon, Jul 7, 2008 at 10:46 AM, Emmanuel Favre-Nicolin
[EMAIL PROTECTED] wrote:
Hi,
I'd like to know if someone already succesfully installed sippyskype.
Here on gentoo, I'm starting to build necessary stuff.
1) mjsip_1.6
I used :
http://blogimg.chinaunix.net/blog/upfile/070801012614.pdf
Julio Arruda wrote:
I would assume this would be against the TermsConditions/AUP of a VOIP
provider..
Wasn't/isn't that kind of the point?
--
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (706)
On Jul 7, 2008, at 7:46 AM, Emmanuel Favre-Nicolin wrote:
Hi,
I'd like to know if someone already succesfully installed sippyskype.
Here on gentoo, I'm starting to build necessary stuff.
I've got sippyskype running on CentOS 5.1.
You shouldn't have to build anything, the application comes
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Klaverstyn, David C wrote:
Hi All,
I was under the impression that I found a WEB site about two years or so
ago that allowed Asterisk users to place free calls between each other
that used up users un-used minutes/calls. I though the site
The author is very responsive to fix submissions. He only has a
SPA3102 and a windows machine to code for, so offering space on an
asterisk box could go a long way.
--
Eric Chamberlain
Founder
RF.com
http://RF.com/
Eric,
I have a dual proc AMD server that he can use as a sandbox if
-- Channel will hangup at 2008-07-08 02:53:51 UTC.
-- Executing [EMAIL PROTECTED]:4] Set(SIP/1XX-007321b0,
FAXFILE=/var/spool/asterisk-fax/20080707-195351-1xx.tif) in
new stack
-- Executing [EMAIL PROTECTED]:5] RxFAX(SIP/1XX-007321b0,
/var/spool/asterisk-fax
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