Quoting Zed [EMAIL PROTECTED]:
In case anyone is wondering its the same in Ontario with Bell Canada,
other cell carriers, and 289 overlaid on 905 and 647 overlaid on 416,
and 416 split to 905 some time ago as well. Some much for the old rule
of every area code had 1 or 0 as the middle
Alvaro, can you post the patch in a public place and post the URL
here? It might be a good idea to contact steve underwood to see what
he has to say about such a patch.
Regards,
On 7/18/07, Alvaro Parres [EMAIL PROTECTED] wrote:
Carlos:
Only for check do this change:
Thanks for the info.
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asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
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Arun Kumar wrote:
Hi,
Congestion() if no of calls in this group are more then 3. But my
provider says he is not getting any busy signal from my side and he
says for all incoming numbers (30) he is getting back only one number
from asterisk box(4340).
exten = 4340,16,Congestion()
Try
Kevin Kiely wrote:
app_valetparking listed here http://www.freeswitch.org/asterisk_stuff/
Indicates support for Asterisk 1.4. The documentation listed suggests an
install like so:
cd /usr/src/asterisk
cp contrib/scripts/astxs /usr/bin/
cd apps
wget http://www.bkw.org/app_valetparking.c
equis software wrote:
I'm using Queue app with Asterisk 1.4.6
It was working 5 days without problems and then it crash.
When I did #gdb asterisk core.xxx
I see...
snip
There is some weird stuff in that backtrace which makes me think it is not
accurate. You have to build without
Hello,
I
Check this page:
http://www.asterisk.net.au/general/1/
It's very interesting
Best Regards
Carlos Rojas
On 7/18/07, Dmytro Mishchenko [EMAIL PROTECTED] wrote:
Tim Reimers wrote:
Hi -
I need to configure Asterisk (Trixbox 2.2) and my ATA-186 with both
ports.
I need to be able
yonoko molomo wrote:
[Jul 17 11:19:22] WARNING[23645]: src/chan_h323.c:1044
ooh323_indicate: Don't know how to indicate condition -1 on ooh323c_1
[Jul 17 11:19:25] NOTICE[23645]: rtp.c:783 process_rfc3389: Comfort
noise support incomplete in Asterisk (RFC 3389). Please turn off on
client if
On Wed, Jul 18, 2007 at 12:44:29AM +0200, Thomas Winter wrote:
Hi,
compile and load of modules works fine.
After ztcfg I can see
.
.
Changing signalling on channel 1 from Unused to Clear channel
Changing signalling on channel 2 from Unused to Clear channel
Changing signalling on
I've included my jabber.conf below. I'm betting the following errors:
[Jul 18 21:05:22] ERROR[28166]: res_jabber.c:609 aji_act_hook: JABBER:
Node Error
[Jul 18 21:05:22] WARNING[28166]: res_jabber.c:1537 aji_recv_loop:
JABBER: Got hook event.
jabber test
[Jul 18 21:04:16] WARNING[32691]:
From memory, Flash() followed by SendDTMF.
PaulH
On Wed, 2007-07-18 at 16:14 -0500, Jay Moore wrote:
Greetings, List.
I have my Asterisk box setup with 8 Centrex lines that were left over
from our old PBX system. My boss is asking me to set up Asterisk so
that he can flash hook and
For some Digium cards maybe flash the hook and then *0 -- try that.
on Thursday 07/19/2007 Paul Hales([EMAIL PROTECTED]) wrote
From memory, Flash() followed by SendDTMF.
PaulH
On Wed, 2007-07-18 at 16:14 -0500, Jay Moore wrote:
Greetings, List.
I have my Asterisk box
I have received mail from Amazon touting this book that will soon be
available.
Know anything about the book or it's authors? It's a little pricey.
Here is the blurb:
Asterisk Cookbook (Paperback)
by Jim Van Meggelen (Author), Leif Madsen (Author), Kristian Kielhofner
(Author), John Todd
Hi,
Your invite is going with ulaw and alaw.
need to check that what are the entries of codecs in your sip.conf, have you
allowed there ulaw and alaw or not, and next thing is if your gateway
accepting, these codecs or not.
Keshav
laurent schweizer [EMAIL PROTECTED] wrote: Hello,
I have a
What problems are you facing ?
- Original Message -
From: Al lists
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Wednesday, July 18, 2007 7:00 AM
Subject: [asterisk-users] AudioCodec MP114
Hi list,
I'm trying to use an AudioCodec Mp114, 4 FXO Media
Thank for answer
I got it you point. i want to give u a idea of my setup i have avaya
Voip and i have intergrate asterisk with avaya through mediant 2000 E1 gateway
[Asterisk]sip trunk---[Mediant2k]-E1[Avaya]
now i have configured second E1 on mediant 2000 for
On 17 Jul 2007, at 11:26, Steve Kennedy wrote:
On Tue, Jul 17, 2007 at 11:56:35AM +0200, Anselm Martin Hoffmeister
wrote:
Am Donnerstag, den 12.07.2007, 16:57 -0700 schrieb Russ McBride:
Newbie question(s):
From what I can determine it sounds like the SMS messaging isn't as
robust as
Hi,
I just spoke with my telco about a problem I have with some zap channels
getting stuck in PRI flags: Resetting when we have a heavy load (lots
of calls). The technican I spoke with told me that this is most likely
because asterisk says the zap channel should be exclusive and this
causes
Hi,
I have a SIP phone which does not natively support SIP transfers (REFER
etc...). So far all that is possible is to enable blind transfers using
the t and T arguments in Dial from the # DTMF key. The phone has an R
button on it and this can be setup to either send an RFC2833 hook flash
Hello List,
I just have a query is it possible to have 2 or more telephone
number mapped to the same E1 line and if so will the TE120P card pick up
the last 4 digits of each number - as it is currently doing for the one?
--
Kind Regards
Etienne Pretorius
Hi All,
Does anyone have an example dial-plan and/or a way to use the queues to
access outgoing calls on Zap channels when all the channels are congested?
e.g. I have 20 users, who will be accessing Zap/g1 .. which has 3 channels
I have another 5 users who will be accessing Zap/g2 .. which
Hi,
did anybody manage to configure Asterisk (1.4.8) with imap voicemail
storage together with an Microsoft Exchange Server (2003)?
I can connect my asterisk to a local dovecot imap server without problems.
But if I change the settings to our exchange server I can´t connect...
Here is my
On 18/lug/07, at 12:31, Etienne Pretorius wrote:
Hello List,
I just have a query is it possible to have 2 or more telephone
number mapped to the same E1 line and if so will the TE120P card
pick up the last 4 digits of each number - as it is currently doing
for the one?
Sure.
--
Hi,
Does anyone know how to change the channel identification on a PRI line
on our asterisk from 'Exclusive' to 'Preferred' or 'Negotiation'? Is
this even possible?
Regards,
Jan
___
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On Wed, 2007-07-18 at 12:31 +0200, Etienne Pretorius wrote:
I just have a query is it possible to have 2 or more telephone
number mapped to the same E1 line and if so will the TE120P card pick
up the last 4 digits of each number - as it is currently doing for the
one?
Yes, Asterisk will
Hello again,
Sorry for replying to myself, but i forgot to include the asterisk cli
output:
Asterisk Ready.
*CLI [Jul 18 15:32:47] ERROR[22690]: app_voicemail.c:8570 mm_log: IMAP
Error: Can't do /authuser with this server
[Jul 18 15:32:47] ERROR[22690]: app_voicemail.c:4674 init_mailstream:
Could you send please your unicall.conf file
Thanks.
It appers to be a problem with de ANI digits you want to recive.
On 7/17/07, Carlos Chavez [EMAIL PROTECTED] wrote:
On Tue, 2007-07-17 at 19:30 -0500, Moises Silva wrote:
In order to help you I need testcall traces, with max level of
Dovid B wrote:
2) If you hav been following the list and the issue you will notice
that the problem is for a lot of users that have their servers out
side of the US. The problems started when digium moved over from easy
news to api-digital. The reason why you probably have no issue is
I didn't paste the actual etxensions.conf entry -- there are quotes in
the file itself.
Any other ideas?
Martin Smith, Systems Developer
[EMAIL PROTECTED]
Bureau of Economic and Business Research
University of Florida
(352) 392-0171 Ext. 221
-Original Message-
From: [EMAIL
Dear all
I have beginer in Voip and i have configured Asterisk server
with 100 IP SIP phone ( SNOM ) everything is fine but problem is how to
transfer call from one user to other means i call to some one and then someone
want to transfer call to another person how it is
Paul Hayes wrote:
However, Asterisk ignores the hook flash messages and I can't find
anyway of getting it to treat the hook flash message in the same way as
a # being sent.
The only information I can find relates to detecting or sending hook
flashes on zap channels.
This is because
Hi,
I'm using Asterisk-1.4.7.1.
Everything was working fine.
Now I'm trying to Install Asterisk-addons-1.4.2.
The procedure I followed is as...
# cd asterisk-addons-1.4.2
#./configure
#make menuselect
#make
#make install
Everything is going fine except make install. I've tried many times, but
Dear all
I am going to implement big setup of asterisk base PBX
arround 200 SIP hardware Phone so what kind of setup would be best means what
kind of hardware i have single PRI on asterisk how much proccession i need to
run this kind of setup and one more thing is i use SER
I have this problem on a te110p. On random basis the calls are
disconnected by asterisk. When this happens asterisk logs:
Jul 18 12:52:10 DEBUG[4668] channel.c: Got a FRAME_CONTROL (5) frame on channel
Zap/13-1
Jul 18 12:52:10 DEBUG[4668] channel.c: Bridge stops bridging channels
Dear all
I have one more question about codec what codec i use for LAN
setup G.729 or Alaw which is best for LAN setup caz some people told me G.729
is use for wan link not for lan caz it is cost effective so can anyone suggest
me best codec for asterisk and SIP phone
Rgds
On Wed, 2007-07-18 at 07:12 -0700, satish patel wrote:
I have one more question about codec what codec i use
for LAN setup G.729 or Alaw which is best for LAN setup caz some
people told me G.729 is use for wan link not for lan caz it is cost
effective so can anyone suggest me
not getting any calls in or out.
practically nothing works.
beisde this, any other media Gateway ?
I dont underestand why Digium is not making one.
On 7/18/07, Dovid B [EMAIL PROTECTED] wrote:
What problems are you facing ?
- Original Message -
*From:* Al lists [EMAIL PROTECTED]
issue is got solved by moving to another pri card and now congestion works
fine with my ISP.
thanks all.
On 7/18/07, Andrew Joakimsen [EMAIL PROTECTED] wrote:
On 7/17/07, Jared Smith [EMAIL PROTECTED] wrote:
On Tue, 2007-07-17 at 12:52 -0400, Andrew Joakimsen wrote:
I did a quick test.
G711 is preferred if you wont face any bandwith limitation.
That is why g729 is used on wan links.
Voice quality should be better than g729 ans also less cpu load for
asterisk.
On 7/18/07, satish patel [EMAIL PROTECTED] wrote:
Dear all
I have one more question about codec
What is the best CODEC/Format combination to use?
I've got several * boxes setup on a lan that are IVR servers. All the
prompts are in GSM so I was using GSM thinking that it would prevent
transcoding between the prompts and the voice channel. Is this an accurate
assumption? Is there a better
Try looking at the CLI output of Asterisk as well as set up a sys log server.
Set the Verbosity level on the Audiocodes to 5 and look at the output of the
Audiocodes box on the syslog server.
- Original Message -
From: Al lists
To: Asterisk Users Mailing List - Non-Commercial
This is a bug. Search for the file and move it over manually.
- Original Message -
From: Keshav K.
To: Asterisk-users Digium
Sent: Wednesday, July 18, 2007 5:09 PM
Subject: [asterisk-users] Issue in insatlling addons-1.4.2
Hi,
I'm using Asterisk-1.4.7.1.
Everything was
On Wed, 18 Jul 2007, satish patel wrote:
Dear all
I have beginer in Voip and i have configured Asterisk
server with 100 IP SIP phone ( SNOM ) everything is fine but problem is
how to transfer call from one user to other means i call to some one and
then someone want to
On Wed, 18 Jul 2007, satish patel wrote:
Dear all
I am going to implement big setup of asterisk base
PBX arround 200 SIP hardware Phone so what kind of setup would be best
means what kind of hardware i have single PRI on asterisk how much
proccession i need to run this
Hi,
To use call tranfer you have to make entry in extension.conf...
exten = _7.,1,Dial(SIP/${EXTEN},20,Ttr)
then in feature.conf
[featuremap]
blindxfer = #8 ; Blind transfer (default is #)
;disconnect = *0 ; Disconnect (default is *)
;automon = *1 ;
I'm having a problem building Asterisk 1.2.22. It fails in
codecs/codec_zap.c on codec_zap.c is revision 62173. The OS is FC4.
Here's the error. Can anyone help me with this?
gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes
-Wmissing-declarations -g3 -Iinclude -I../include -D_REENTRANT
There is no libchan_h323.so.1.0.1 file in libs...
See here all the files of .libs
asterisk-ooh323c]# ls -l .libs/
total 3732
lrwxrwxrwx 1 root root 18 Jul 18 19:21 libchan_h323 - libchan_h323.1.0.1
lrwxrwxrwx 1 root root 18 Jul 18 19:21 libchan_h323.1 -
libchan_h323.1.0.1
Something changed in the final linking of the channel, and it now
produces libchan_h323.1.0.1 instead of libchan_h323.so.1.0.1
Either edit the Makefile to copy libchan_h323.1.0.1, or manually
copy that file...
Dan
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Keshav
Has anyone tried to do a script to pickup an ITSP voicemail.
Lesnet provides an option for an overflow mailbox in the event a caller can get
to my * box.
I'd like my * to poll it and dump any messages found into my general mailbox
Any ideas
Similarly, a telco mailbox. It at least has the
I wonder what version of Zaptel you are using -- sounds like you have
not installed a new version or you are using an older one.
on Wednesday 07/18/2007 Daryl Jones([EMAIL PROTECTED]) wrote
I'm having a problem building Asterisk 1.2.22. It fails in
codecs/codec_zap.c on codec_zap.c is
On Wed, 2007-07-18 at 08:10 -0500, Alvaro Parres wrote:
Could you send please your unicall.conf file
Thanks.
It appers to be a problem with de ANI digits you want to recive.
loglevel=1
protocolclass=mfcr2
protocolvariant=mx,10,4
protocolend=cpe
language=es
usecallerid=yes
On Wed, 2007-07-18 at 08:10 -0500, Alvaro Parres wrote:
Could you send please your unicall.conf file
Thanks.
It appers to be a problem with de ANI digits you want to recive.
Also Nextel never sends CallerID. When someone calls me from a Nextel
phone to my cell or to my
Hi all,
I've just installed again my Asterisk using Xorcom repositories. I can
make extensions, but when using any extension i want to dial anything,
I got 404 not found using Xlite.
Any ideas of what can be happening?
Regards,
--
Diego Quintana a.k.a. RouterMaN
Ingeniero de las
On Wed, 2007-07-18 at 12:07 -0500, Diego Quintana Cruz wrote:
Hi all,
I've just installed again my Asterisk using Xorcom repositories. I can
make extensions, but when using any extension i want to dial anything,
I got 404 not found using Xlite.
My guess is that your user (or friend) account
Hello everyone,
I'm having some problems with receiving DTMF on my incoming PRI. It seems
that incoming DTMF is changed so that it is being generated at a slower
speed. This is not a problem for voice calls, but some of the traffic on
the PRI consists of alarm control panels that transmit
2007/7/18, Jared Smith [EMAIL PROTECTED]:
On Wed, 2007-07-18 at 12:07 -0500, Diego Quintana Cruz wrote:
Hi all,
I've just installed again my Asterisk using Xorcom repositories. I can
make extensions, but when using any extension i want to dial anything,
I got 404 not found using Xlite.
2007/7/18, Diego Quintana Cruz [EMAIL PROTECTED]:
2007/7/18, Jared Smith [EMAIL PROTECTED]:
On Wed, 2007-07-18 at 12:07 -0500, Diego Quintana Cruz wrote:
Hi all,
I've just installed again my Asterisk using Xorcom repositories. I can
make extensions, but when using any extension i want
On 7/18/07, Dave Bour [EMAIL PROTECTED] wrote:
Has anyone tried to do a script to pickup an ITSP voicemail.
Lesnet provides an option for an overflow mailbox in the event a caller can
get to my * box.
I'd like my * to poll it and dump any messages found into my general
mailbox
Any ideas
Is there a way to hang up on a sip channel. One of my phones is saying
it's busy while it's not (even after rebooting it).
I logged into asterisk, and did a sip show channel 232, and sure enough
it thinks it's on a call.
How can I force it to close?
Carlos:
Only for check do this change:
protocolvariant=mx,10,4
for
protocolvariant=mx,0,4
If it's works, contact me and i will send you a patch for libmfcr.c
Thanks.
Carlos:
Has el cambio que te pido arriva, para revisar si es lo del caller ID.
Casi estoy seguro nosotros en
On Wed, 2007-07-18 at 15:39 -0400, Dan Casey wrote:
How can I force it to close?
Type soft hangup SIP/channel-name at the Asterisk CLI. You can get
the channel name from the output of sip show channels.
--
Jared Smith
Community Relations Manager
Digium, Inc.
soft hangup channelname
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dan Casey
Sent: Wednesday, July 18, 2007 3:39 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Force SIP hang up.
Is there a way to hang up on
Both telco and ITSP are dial in and play message, delete, etc via various key
strokes.
Dave Bour
Desktop Solution Center
905.381.0077
[EMAIL PROTECTED]
For those who just want it to work...
Giving you complete IT peace of mind.
(Sent via Blackberry - hence message may be shorter than my
Correct. zaptel-1.2.12 is currently installed. I plan to install
zaptel-1.2.19 as part of this upgrade. zaptel-1.2.19 compiled clean, but
has not been installed yet.
John covici wrote:
I wonder what version of Zaptel you are using -- sounds like you have
not installed a new version or you
Hi, All
I have asterisk installed behind non-symmetric NAT, so I have NAT traversal
issues.
How can I use STUN or TURN to register with the other end? Or Asterisk
doesn't support it?
Thanks
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But asterisk will not compile till you install the correct version of
zaptel.
on Wednesday 07/18/2007 Daryl Jones([EMAIL PROTECTED]) wrote
Correct. zaptel-1.2.12 is currently installed. I plan to install
zaptel-1.2.19 as part of this upgrade. zaptel-1.2.19 compiled clean, but
has not
I've been evaluating Asterisk for a while, and things seem to be
going very well. The issue of redundancy and automatic fail-over is
now on my mind. I searched the archives and googled for solutions,
but didn't really come up with much.
We'll be using queues (modified), which precludes some
That's what I needed to know. Thanks!
John covici wrote:
But asterisk will not compile till you install the correct version of
zaptel.
on Wednesday 07/18/2007 Daryl Jones([EMAIL PROTECTED]) wrote
Correct. zaptel-1.2.12 is currently installed. I plan to install
zaptel-1.2.19 as part
Asterisk Users,
I have Asterisk PBX System running at my work. The system is working
great. Currently, I have Broadvoice as my sip provider and I am not
completely satisfy with their service. Broadvoice only allows 2
simultaneous calls, which hinders my company's communications ability.
Greetings, List.
I have my Asterisk box setup with 8 Centrex lines that were left over
from our old PBX system. My boss is asking me to set up Asterisk so
that he can flash hook and make an outgoing call on the same line to
have a 3 way call.
This is what he wants to do:
1) Incoming call on
I know that Voicepulse can do this. By default the offer 4 channels, but you
can buy the other two or how many others you need as well.
http://connect.voicepulse.com
Tom
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of John Meksavan
Sent: Wednesday, July
For termination try voipjet.com. Lots of people seem to only use them as a
back up. I use them as primary and if it fails I have it set to teliax.
For termination and or origination try teliax.com
I have recently started testing jivetel.com. So far so good but I have only
been testing for 2
Tim Reimers wrote:
Hi -
I need to configure Asterisk (Trixbox 2.2) and my ATA-186 with both ports.
I need to be able to call one port from the other-- the idea is to have
two phones in two different locations that _can_ call each other.
So, in reading the Asterisk Wiki and
I've got a 778 DID for vancouver, but don't know if it will be a local call
if dialed 604 and vice versa.
What are the different area codes in Vancouver and why its easier to get 778
DID than 604?
--
Zeeshan A Zakaria
___
--Bandwidth and Colocation
I'm trying to add MySQL CDR recording in Asterisk 1.4.6
I'm following the instructions posted here:
http://www.voip-info.org/wiki-Asterisk+cdr+mysql
but after running make, the cdr_addon_mysql.so is not created. I
don't get any compile errors. In fact it just seems to skip the
compile
On Wed, 2007-07-18 at 19:41 -0400, hugolivude wrote:
I'm trying to add MySQL CDR recording in Asterisk 1.4.6
I'm following the instructions posted here:
http://www.voip-info.org/wiki-Asterisk+cdr+mysql
but after running make, the cdr_addon_mysql.so is not created. I
don't get any
On 7/18/07, Tom Moore [EMAIL PROTECTED] wrote:
I know that Voicepulse can do this. By default the offer 4 channels, but you
can buy the other two or how many others you need as well.
http://connect.voicepulse.com
Tom
VoicePulse is very bad IMO. They charge you for those extra channels
Check here: http://www.localcallingguide.com/
AR
On 7/18/07, Zeeshan Zakaria [EMAIL PROTECTED] wrote:
I've got a 778 DID for vancouver, but don't know if it will be a local
call if dialed 604 and vice versa.
What are the different area codes in Vancouver and why its easier to get
778 DID
On 7/18/07, satish patel [EMAIL PROTECTED] wrote:
Dear all
I have beginer in Voip and i have configured Asterisk
server with 100 IP SIP phone ( SNOM ) everything is fine but problem is how
to transfer call from one user to other means i call to some one and then
someone want
On 7/18/07, Dave Bour [EMAIL PROTECTED] wrote:
Both telco and ITSP are dial in and play message, delete, etc via various
key strokes.
I am not aware of any way to do that. I was thinking perhaps your
provider offered some sort of API for message retrieval. If your
provider is SIP or IAX you
Zeeshan Zakaria wrote:
I've got a 778 DID for vancouver, but don't know if it will be a local
call if dialed 604 and vice versa.
What are the different area codes in Vancouver and why its easier to
get 778 DID than 604?
Yes they are both the same calling area. The 778 area code is an
On 7/18/07, Jay Moore [EMAIL PROTECTED] wrote:
Greetings, List.
I have my Asterisk box setup with 8 Centrex lines that were left over
from our old PBX system. My boss is asking me to set up Asterisk so
that he can flash hook and make an outgoing call on the same line to
have a 3 way call.
On 7/18/07, Zeeshan Zakaria [EMAIL PROTECTED] wrote:
I've got a 778 DID for vancouver, but don't know if it will be a local call
if dialed 604 and vice versa.
What are the different area codes in Vancouver and why its easier to get 778
DID than 604?
Answer to the first part is YES. 778 is
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