In the [general] section, put:
register = 8159093010:[EMAIL PROTECTED]
Then add a SIP peer for the outbound proxy. Something like:
[essex1_outbound]
fromdomain=proxy.essex1.com
host=proxy.essex1.com
port=5060
insecure=very
username=8159093010
secret=X
type=peer
qualify=no
canreinvite=no
Linksys SPA 2102. No issues at all. Period.
Regards,
Mindaugas Kezys
http://www.kolmisoft.com
MOR PRO - Advanced Billing for Asterisk PBX
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Adam Moffett
Sent: Wednesday, February 20, 2008 11:26 PM
To:
Cyber-Telecom's CT-V372 is same box as PorTech MV-372 but with more advanced
firmware. It supports more functions, such as SMS sending.
Regards,
Mindaugas Kezys
http://www.kolmisoft.com
MOR PRO - Advanced Billing for Asterisk PBX
-Original Message-
From: [EMAIL PROTECTED]
Hi All,
I would just like to clarify the requirements of the h323 channel within
asterisk.
Can I use a recent edition of PTLib and OpenH323, for example, the
editions located at OpenH323+:
http://www.h323plus.org/source/
OpenH323+ v1.20.2
PTLib v2.0.1
Or do I need to use the versions at the
A while back i had asked about possible replacements for snom 360 phones
that were breaking and causing issues and we all discussed the problems
we had with the 360s and some suggestions were made but the new polycom
phones had just hit the market and not many people were able to comment
on them.
Hello,
can Asterisk be used in a 3PCC scenario as described in RFC:
ftp://ftp.rfc-editor.org/in-notes/rfc3725.txt
I'm not meaning using Asterisk as the controller, I mean Asterisk be
controlled by a 3rd party Back to Back User Agent.
In this case can Asterisk translate Sip into iax and hiding
Of the three ATAs I've got (Linksys PAP2-NA, Sipura SPA-2000 and
SPA-3000) the Linksys PAP2-NA is the best of the bunch, even though the
SPA-2000 is supposedly cut from the same mould.
For the most part, you set 'em and forget 'em. Most of the time when I
have a problem with a phone connected to
I am personally Waiting u :) -
Thanks in advance.
Regards
Bilal
---
I may have found a solution to why this problem is
happening to me. All
my
IAX trunks are up and working and have been for over a
day now. If
there are
still up and running with no problems in a week I
This can help (script for Debian):
apt-get install flex bison
#dirty hack to prevent error from missing file
cd /usr/include/linux
touch compiler.h
#PWLIB
cd /usr/src
wget
http://kent.dl.sourceforge.net/sourceforge/openh323/pwlib-v1_10_0-src-tar.gz
tar zxvf pwlib-v1_10_0-src-tar.gz
cd
We do:
in modules.conf:
noload = pbx_ael.so
noload = pbx_dundi.so
noload = res_config_pgsql.so
noload = res_smdi.so
in extensions.conf delete every context [default], [demo], whatever
in sip.conf, iax.conf delete all peer/users if any
Regards,
Mindaugas Kezys
http://www.kolmisoft.com
MOR PRO
Hello,
To compile chan_h323 as is distributed you need to download OpenH323
v1.18.0 and PwLib v1.10.0 from:
http://www.voxgratia.org
Some months ago I had made a patch to compile the 1.4.x version and the
trunk version (which evolved to 1.6.x) with H323+.
Sadly, the patch was not included in
Hi,
Thanks for the information, I will keep this for reference.
Thanks
Bruce
Mindaugas Kezys wrote:
This can help (script for Debian):
apt-get install flex bison
#dirty hack to prevent error from missing file
cd /usr/include/linux
touch compiler.h
#PWLIB
cd /usr/src
wget
Hi,
Thank you for the details of which versions to get. I will be building
these two versions on Solaris to test chan_h323.
Did your patch for building with OpenH323+ make it into the 1.4 edition
of Asterisk?
Thanks
Bruce
Vlasis Hatzistavrou (KTI) wrote:
Hello,
To compile chan_h323 as is
Hi List,
Recently I tried sending sms using app_sms (hardware TDM400P) in Singapore
with land line telco provider singtel
it worked fine and can send sms in Latin characters 7-bits/8-bits
but I am unable to send Unicode (UCS-2 or 16-bits) sms in Arabic or Chinese.
the problem is that my
On Thu, 21 Feb 2008 15:00:15 +1100, Paul Hales
[EMAIL PROTECTED] wrote:
Head off into /etc/asterisk/modules.conf and add some 'noload' lines.
Ah, makes sense. Asterisk loads everything, and must be told
explicitely _not_ to load something :-)
Is there a comprehensive list that explains what each
Adam Moffett wrote:
In all seriousness, my requirements were a little silly. A Cisco router
can fail just as a netgear router can. But I think we would find Cisco
failures to be statistically less likely.
I also think we can agree that not all devices of a certain type are
created
first off I anwered you to use vi and you complained showing me cat.
then for your next question about ehat each module does. show module
in asterisk in combination with show application as well as a peak at
the source should give you a clue.
also the module names are quite descriptive.
On
since if this email might contain confidential info, in which case I
must not review it. I can't really reaed it since it implies that I
read the disclaimer before the body of the email, and as such as long
as the disclaimer is in it I must NEVER Aread the email.
On 2/21/08, Michael J. Liberatore
Hi,
I would like to configure asterisk to allow INVITE for hold to pass
through it and not provide music on hold by itself. Can anyone help me out
here?
Regards,
Mayur
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I have been looking for one of the many USB to ISDN products listed on this
page,
http://www.nslu2-linux.org/wiki/OpenSlug/Asterisk
But I have not been able to find a single item for sale anywhere in the US,
has anyone seen where I can find a couple of these?
Thanks
Michael
Al lists wrote:
Has anyone checked asterisk with check_udp plug in?
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asterisk-users mailing list
To
autoload=yes says to load everything, so you either need to change it
to no and then add load statements for every module you need, or leave
it as yes and then add noload for everything you don't need.
Vincent wrote:
On Wed, 20 Feb 2008 21:44:30 -0500, C F [EMAIL PROTECTED] wrote:
vi
Quoting SIP [EMAIL PROTECTED]:
Adam Moffett wrote:
In all seriousness, my requirements were a little silly. A Cisco router
can fail just as a netgear router can. But I think we would find Cisco
failures to be statistically less likely.
I also think we can agree that not all devices of a
This may be a dumb question, but I have never done menus, how do I link
the below up to my phone number? For example, right now I route calls
to the SIP phone like so:
[ipcomms]
include = default
exten = 211212, 1, Dial(SIP/6000,20,tr)
where ipcomms is my context from sip.conf for
Mitchell Jackson wrote:
Hello! I am having problems figuring out how to do something, and any
help would be much appreciated.
I would like to use the manager API to take an existing call on a
specific SIP extension, dial and conference in a third party.
From what I can tell, the way
Hello Bruce,
Bruce McAlister wrote:
Did your patch for building with OpenH323+ make it into the 1.4 edition
of Asterisk?
No, it didn't as it was considered a new feature and by Digium's policy
new features can only be added in the trunk versions.
The strange thing is that I added it in
Jon, did you ever discover a solution to your problem. I'm in the same boat.
On Sun, Dec 2, 2007 at 1:22 PM, Jonathan C. Bailey
[EMAIL PROTECTED] wrote:
I'm having (I think) timing issues in relation to bridged T1-T1 calls via
dynamic spans. Fax calls are intermittently working, but voice is
SIP wrote:
Adam Moffett wrote:
In all seriousness, my requirements were a little silly. A Cisco router
can fail just as a netgear router can. But I think we would find Cisco
failures to be statistically less likely.
I also think we can agree that not all devices of a certain type are
Re: Contents of asterisk-users digest
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asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Does anyone have a audio file they would be willing to share for on hold
music?
I am looking for something like the old norstar beep every few seconds.
I tried 3 seconds silence, beep.wav, beep.wav. But it just didn't sound
right. I need one that has a softer beep.
Thanks!
--
***
Forrest
Hi list,
I'm having problems transferring certain calls made by the attendant between
the PSTN and to an internal extension. Although, transfers between the
majority of the calls ends successfully.
Debugin this, I've found that calls made to certain numbers (Telephony
Providers), aren't detected
I am having issues with SendDTMF on an Asterisk box using Asterisk 1.4.18 and
Zaptel 1.4.8. As far as I can see, the issues seem to be related to echo
cancellation. The box has a TDM2400p installed and is using the HPEC echo
canceller.
The problems occur when I attempt to do an outcall. The
Hello everyone,
I have many Asterisk clients registered to an OpenSER proxy.
Sometimes (for reasons unknown) the 407 Proxy Authentication Required
sent by OpenSER to Asterisk is not received by Asterisk on the client,
causing the call to fail. No other SIP messages or other IP traffic
seems to
On Thursday 21 February 2008 08:44:11 John Von Essen wrote:
This may be a dumb question, but I have never done menus, how do I link
the below up to my phone number? For example, right now I route calls
to the SIP phone like so:
[ipcomms]
include = default
exten = 211212, 1,
Any Linksys ATA is best value for money. I have used SPA AND PAP 2 series.
I have also CISCO ATA in office and still no problems for almost 2 years.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anthony Francis
Sent: Thursday, February 21, 2008 5:19 PM
Hi,
Which echo-canceler shall I pick for Digium B410P ?
Is HPEC relevant ? Reading from its datasheet, it seems related to analog
cards.
Regards
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asterisk-users mailing list
To
Hi All,
I have setup 2 trunks using 2 different voip providers using sip.
the first one i have no problem calling inbound then redirected to an
IVR, i can hear the IVR.
the second one has issues, inbound works going to IVR as i can see it on
the CLI, but i don't hear anything. i tried
Raúl,
From your conf file I guess the CO provide reversal polarity for answer
supervision. Verify if for those numbers the CO revert the line
polarity when callee answer.
callprogress=no is a good test too.
Jorge
Raúl Gómez C. wrote:
Hi list,
I'm having problems transferring certain calls
I put that in, but it appears that it is trying to contact the private IP
address of their SIP server. I have successfully registered to this server
from over the public Internet using an Innomedia ATA.
[Feb 21 11:49:18] NOTICE[4608]: chan_sip.c:7364 sip_reg_timeout:--
Registration for
The HWEC, not software.
-Darren
From: [EMAIL PROTECTED] on behalf of Olivier
Sent: Thu 2/21/2008 12:37 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Which echo-can for Digium B410P ?
Hi,
Which echo-canceler shall
Kevin P. Fleming wrote:
I've just located an E400P from our graveyard of old cards... if it
works, I'll be able to solve this problem in the morning.
This has been fixed in revision 3863 of the 1.4 branch; it's a one line
fix that you should be able to easily apply to existing Zaptel source
On Thu, Feb 21, 2008 at 12:40 PM, Ron [EMAIL PROTECTED] wrote:
Hi All,
I have setup 2 trunks using 2 different voip providers using sip.
the first one i have no problem calling inbound then redirected to an
IVR, i can hear the IVR.
the second one has issues, inbound works going to IVR
Will this work to match any number from the 770,404, or 678 area codes?
_[404|770|678]NXX
If this won't work, is there a pattern that will do this?
Yours,
Michael Munger, dCAP
404-438-2128
[EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
Attachment encrypted? click here
=) in new stack
-- Executing Macro(SIP/23456789-0821de80, record-enable|300|IN) in new
stack
-- Executing GotoIf(SIP/23456789-0821de80, 0?2:4) in new stack
-- Goto (macro-record-enable,s,4)
-- Executing AGI(SIP/23456789-0821de80,
recordingcheck|20080221-175654|1203645383.3) in new
I'm a huge fan of the Linksys SPA-942s for users. They run around $125, are
pretty straightforward to manage via TFTP, and work really well with
Asterisk.
On Thu, Feb 21, 2008 at 4:07 AM, Michael J. Liberatore
[EMAIL PROTECTED] wrote:
A while back i had asked about possible replacements for
Delete extensions.ael too, unless you're using AEL instead of the dialplan
Mindaugas Kezys wrote:
We do:
in modules.conf:
noload = pbx_ael.so
noload = pbx_dundi.so
noload = res_config_pgsql.so
noload = res_smdi.so
in extensions.conf delete every context [default], [demo], whatever
in
Michael,
I'm very interested in helping you resolve any and all trouble you have with
Digium products. The experience you describe with our products not typical, so
I'd like to address them specifically.
My comments are inline:
- Michael J. Liberatore [EMAIL PROTECTED] wrote:
I have
On Thu, 2008-02-21 at 12:44 -0500, Michael Munger wrote:
Will this work to match any number from the 770,404, or 678 area
codes?
_[404|770|678]NXX
If this won’t work, is there a pattern that will do this?
No, it won't work, there's no '|' for alternative matches, and
[746][704][048]
[At 01:21 PM 2/21/2008, you wrote:
On Thu, 2008-02-21 at 12:44 -0500, Michael Munger wrote:
Will this work to match any number from the 770,404, or 678 area
codes?
_[404|770|678]NXX
If this wonât work, is there a pattern that will do this?
No, it
Thanks Jorge, I'll be checking that...
On Fri, Feb 22, 2008 at 1:15 PM, Jorge Mendoza [EMAIL PROTECTED] wrote:
Raúl,
From your conf file I guess the CO provide reversal polarity for answer
supervision. Verify if for those numbers the CO revert the line
polarity when callee answer.
On Thu, 2008-02-21 at 13:34 -0500, Mike Trest - Personal wrote:
[746][704][048]
Nope, that's not going to do exactly what you want either... that
pattern would match a lot of area codes besides the ones you're looking
for. (For example, you could have 7 as the first digit and 0 as the
second
Thank you very much!!!
What was the one line fix?
Also, what file was the problem in? Also, if you know the line number or
function it was in, that would be nice too. I'd do a diff, but I assume
there has been other changes since 1.4.9 was released.
Thanks again!
-- Nick
On Thu, 21 Feb
On Thu, Feb 21, 2008 at 09:23:49AM -0900, Mojo with Horan Company, LLC wrote:
Mindaugas Kezys wrote:
We do:
in modules.conf:
noload = pbx_ael.so
noload = pbx_dundi.so
noload = res_config_pgsql.so
noload = res_smdi.so
Delete extensions.ael too, unless you're using AEL instead of
On Thu, Feb 21, 2008 at 02:17:20PM -0500, Nick Seraphin wrote:
Thank you very much!!!
What was the one line fix?
Also, what file was the problem in? Also, if you know the line number or
function it was in, that would be nice too. I'd do a diff, but I assume
there has been other
The Asterisk.org development team has released Asterisk-addons version 1.4.6.
This releases includes a fix for a build related issue for the OOH323 channel
driver. (issue #9643)
Thank you for your support!
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No that will not work. You would want three exten = lines, one for
each area code.
Michael Munger wrote:
Will this work to match any number from the 770,404, or 678 area codes?
_[404|770|678]NXX
If this won't work, is there a pattern that will do this?
Yours,
That will match the following as well
770
700
740
400
470
670
600
604
608
etc.
You example says:
The first digit can be 7 or 4 or 6. The 2nd digit can be 7 or 0 or 4.
The 3rd digit can be 0 or 4 or 8.
Mike Trest - Personal wrote:
[746][704][048]
[At 01:21 PM 2/21/2008, you wrote:
On
The Asterisk.org development team has released Asterisk-addons version
1.6.0-beta2.
This release contains the following improvement, along with some other minor bug
fixes.
- 11614, Updated app_fax to allow termination and origination of faxes over
T.38
The full list of changes is
The Asterisk.org development team has released version 1.6.0-beta4.
Here are some highlights from the changes, with the associated issue numbers
from bugs.digium.com if an issue was associated with the change.
This release contains the following improvements:
- 12020, a CLI formatting
Jorge,
I think our telco doesn't provide disconnection supervision because I had to
use callprogress, busydetect and busycount in order to properly
disconnect a terminated call (and to avoid the infamous long message in the
voicemail), so I think I can't disable the callprogress option.
I will
I am looking for an ATA like device but instead of VOIP to analog
phone
I want VOIP to low level audio out. Something that looks like a sound
card
output.
I know I can use cheap PC's but that then you have HD's to setup
etc...
HD failures etc...
Anyone know of something like that?
Hi,
Since I upgraded from Asterisk 1.2.18 to 1.4.17 I've been experiencing
high CPU utilization from the chan_sip module. I've notice the more sip
peers I have loaded, the higher the CPU load goes when there are no
active calls. I am currently using a Pentium 4 3.0Ghz with CentOS 4
Kernel
What is it that you think is missing, call comes into incomming, call gets
queued, member [EMAIL PROTECTED] is called and a macro is played to them, they
hit
option 3, MACRO_RESULT gets set to CONTINUE and the call hangs up on the
member while the caller continues on... the caller now though
For those folks who are still using it --
I updated the cid_rewrite script. I noticed that two of the providers
were iffy and one had changed format a little while ago. It's working
again.
http://muware.com/asterisk has the latest (1.2.0)
Enjoy,
-- JM
On Thu, 2008-02-21 at 21:12 +, [EMAIL PROTECTED] wrote:
I currently have 1558 sip peers loaded in
Asterisk and the current CPU load is 10% when no calls are being
processed and no sip registrations.
At first glance, I would think that maybe you have qualify=yes in each
of your SIP peers,
Hi List;
How can I know the needed Zaptel and Kernel versions
for my Asterisk version? Where I can find the
compatibility matrix for such thing?
Regards
Bilal
Looking for last minute shopping deals?
Raúl,
Callprogress is not reliable for call supervision. Sorry.
For maximum reliability with callprogress, the tones and cadences send
by the CO must match every well with the tones plan defined in your
asterisk box. Probably the tones of the other telephone company, where
the answer detection
The Xorcom stuff should be easy enough to find.
PaulH
On Thu, 2008-02-21 at 06:42 -0700, Michael Blood wrote:
I have been looking for one of the many USB to ISDN products listed on
this page,
http://www.nslu2-linux.org/wiki/OpenSlug/Asterisk
But I have not been able to find a
I have questions about AGI.
1. When Using CONTROL STREAM FILE command with all the parameter, I could
not find any way to * or # in the DTMF, it only returns if any digit is
pressed, even if I set forward and rewind digits to BLANK ()
2. When I call out using ZAP, is their a way to find if
bilal ghayyad wrote:
How can I know the needed Zaptel and Kernel versions
for my Asterisk version? Where I can find the
compatibility matrix for such thing?
There is no such thing. If the version of Zaptel you have isn't
compatible with the version of Asterisk you are trying to build,
Mayur wrote:
I would like to configure asterisk to allow INVITE for hold to pass
through it and not provide music on hold by itself. Can anyone help me
out here?
In Asterisk 1.4 this is a configuration option; check the sample
sip.conf file in the configs directory for documentation.
--
Hi all,
Do some one experiencing running jabber applications (jabberstatus...) in
asterisk? I do experinced Asterisk 1.4.18 and wish to start it, however I
got such result.
IBM*CLI help jabber
No such command 'jabber'.
IBM*CLI help jabberstatus
No such command 'jabberstatus'.
Any one can
Tzafrir Cohen wrote:
Delete extensions.ael too, unless you're using AEL instead of the dialplan
extensions.ael is harmless on its own.
It seemed that the default extensions.ael created some demo contexts and
extensions that might befuddle a new user, I could be wrong
I` using chan_h323 on my asterisk-1.4 to receive incomings calls. I need
to set just two codecs to receive this call (g723 and g729), but I`m using
disallow=all
allow=g729
allow=g723.1
In h323.conf, but when I received a call using codec g711 for example,
the call is answered, but doesn`t have
hi
does any body know which version combination of
spandsp/tx_fax/rx_fax will work with * 1.2.24?
i tried different combo. they're either seg fault
during runtime or won't compile.
very frustrated :/
p.s. i know. hylafax/iaxmodem is far more stable. but i have
specific reasons to use rx_fax.
A while back i had asked about possible replacements for snom 360 phones
that were breaking and causing
issues and we all discussed the problems we had with the 360s and some
suggestions were made but the
new polycom phones had just hit the market and not many people were able to
comment on them.
hi,
how to write a advanced dial plan
for example:
dial to a extension(123).if the user didnot pick the call, caller should get a
ivr script(Enter 1 to to dial operator and 2 to go to voicemail)
If caller press 1 it should dial to the operator,else if he dials 2 it should
go to the voicemail
Does anyone have a script that will emulate a normal numeric pager but
send the number to an email address? Also anyone happen to have the
traditional tones used in North America?
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Hi,
It is very strange that following chan_woomera code part gives IP address
44.215.5.41.
static int connect_woomera(int *new_socket, woomera_profile *profile, int
flags)
{
struct sockaddr_in localAddr, remoteAddr;
struct hostent *hp;
struct ast_hostent ahp;
int res = 0;
*new_socket=-1;
Happy coming Spring,
Every week we try to get guests with ideas, products and services you
haven't had time to check out to come and talk about what they're
doing.
Friday, February 22 at 12:00 PM (Eastern US) 9AM PST, 5PM GMT
* Call (724) 444-7444 or sip:[EMAIL PROTECTED]
After the
Hello,
I was never able to get the TE407P card running on a 2.4 Linux kernel.
Using a 2.6 kernel I was able to get it working.
Not really surprising since a lot of companies do not support or even
test on Linux 2.4 any more.
MATT---
On 2/21/08, Kevin P. Fleming [EMAIL PROTECTED] wrote:
bilal
On Thu, Feb 21, 2008 at 7:32 PM, arkda [EMAIL PROTECTED] wrote:
I'm a huge fan of the Linksys SPA-942s for users. They run around $125, are
pretty straightforward to manage via TFTP, and work really well with
Asterisk.
I agree, we've had zero trouble with these. Easy to install and they just
Hi All,
I am trying to build chan_h323 for use with asterisk 1.4.18 on Solaris
10. When I compile asterisk, the build fails at chan_h323 with:
--
chan_h323.c: In function `reload_config':
chan_h323.c:2863: error:
I've done similar notifications in the dialplan.
It would probably look something like this:
exten = s,1,Read(PAGE,enter-phone-number10,10)
exten = s,2,System(/bin/echo Page content: ${PAGE} | /bin/mail -s
Page subject [EMAIL PROTECTED])
-Original Message-
From: [EMAIL PROTECTED]
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