Hi All,
I've got this dialplan:
[macro-callout-intl]
exten = s,1,ResetCDR(w)
exten = s,2,Dial(IAX2/${ARG1}/018${OUTBOUND}||t|L(${OUTTIME}00:6000))
exten = s,3,Goto(s-${DIALSTATUS},1)
exten = s,4,Hangup(19)
exten = s-BUSY,1,NoCDR()
exten = s-BUSY,n,Playback(useris-curntly-busy)
exten =
On Wed, 2010-12-22 at 18:10 +0800, Ron wrote:
Hi All,
I've got this dialplan:
[macro-callout-intl]
exten = s,1,ResetCDR(w)
exten = s,2,Dial(IAX2/${ARG1}/018${OUTBOUND}||t|L(${OUTTIME}00:6000))
exten = s,3,Goto(s-${DIALSTATUS},1)
exten = s,4,Hangup(19)
exten = s-BUSY,1,NoCDR()
exten
Hi All,
Just a little over thought. Sorry if someone already asked about this
before.
Is it possible to put all 16 Ports of E1 in One Asterisk Server ?
And if it's not possible is there any suggestion or alternative for me to
use more than 320 lines of outgoing calls on One Asterisk
On Wed, Dec 22, 2010 at 8:50 AM, Zoel Hairi - Yahoo
zoelha...@yahoo.co.id wrote:
Hi All,
Just a little over thought. Sorry if someone already asked about this
before.
Is it possible to put all 16 Ports of E1 in One Asterisk Server ?
And if it’s not possible is there any suggestion or
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Zoel Hairi -
Yahoo
Sent: Wednesday, December 22, 2010 5:50 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] Maximum E1 Ports on Asterisk ?
Hi
On 21 December 2010 22:06, Tilghman Lesher tilgh...@meg.abyt.es wrote:
On Monday 20 December 2010 14:39:36 Ernie Dunbar wrote:
We have an issue with our Asterisk install where Asterisk produces many
Zombie processes (on the order of several hundred per minute) until
either the Asterisk server
Hello
I have an Asterisk 1.4 server and two XLite softphones, where
Asterisk and the local XLite phone are located in a LAN behind a NAT
router, and the remote XLite phone is located elsewhere on the Net
behind its own NAT router:
http://img252.imageshack.us/img252/3749/asterisknat.png
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Andrew Latham
Sent: Wednesday, December 22, 2010 6:09 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Maximum E1
On 22 December 2010 12:44, Gilles codecompl...@free.fr wrote:
Hello
I have an Asterisk 1.4 server and two XLite softphones, where
Asterisk and the local XLite phone are located in a LAN behind a NAT
router, and the remote XLite phone is located elsewhere on the Net
behind its own NAT
Hello,
you have a typicall nat issue. Asterisk receives messages from the phone
but cannot send any messages back (thats why it tries to resend the 200
ok message 6 times).
try setting qualify=yes to your sip peers config to keep the nat port open.
best regards
stefan
Am 22.12.10 13:44,
On Wed, Dec 22, 2010 at 2:01 AM, Jeroen Eeuwes jeroeneeu...@gmail.com wrote:
Hi Stephen,
_NXXNXX
_NXX
_011.
_911
Of course it can, but it depends on what you want to do when those
numbers are called...
I didn't know about the setvar in the sip.conf and actually I think it
is a
Job Description: Asterisk MySQL Support Engineer
Fast Growing Global Telecoms Company requires a very experienced engineer who
has a variety of skill levels. The role would suit someone who has worked at
switch level and fully understands how calls are to be handled to and from a
VoIP
I am trying to include the ${HANGUPCAUSE} in my mySQL cdr tables. I have a
field called cause_code but it won't write. I belive it is because the
record has already been written by the time I hit the h section of the
code. How might I get this info into the CDR. I need this info for Quality
Hi Nikhil,
Both debug and verbose are set to 20. That's all I got, but as you can see,
for the other types of reasons, the DIALSTATUS got a value (and we see the
events). I'm pretty sure it's a bug.
Michael
On Wed, Dec 22, 2010 at 9:01 AM, Nikhil d.nik...@cem-solutions.net wrote:
Hi
Hi I have tried setting endbeforehexten=yes but still CDR does not get
inserted before h exten. what i tried is setting ResetCDR(w) before the
DEADAGI. Like this:
exten = h,1,ResetCDR(w)
exten = h,2,DEADAGI(get-unqiueid.php)
it seems to work but it's inserting 2 record on the CDR, one with
Hello everyone,
I have noticed thar our dahdi-channels.conf has some repeating directives,
for instance for channel 2 (FXO) we have these settings:
;;; line=2 WCTDM/0/1 FXSKS
signalling=fxs_ks
callerid=asreceived
group=0
context=from-pstn
channel = 2
callerid=
group=
context=default
As you can
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Oguzhan Kayhan
Sent: Wednesday, December 22, 2010 4:11 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] callerid and user
pbx18*CLI module load codec_g729-ast14-gcc4-glibc-pentium3.so
Unable to load module codec_g729-ast14-gcc4-glibc-pentium3.so
Command 'module load codec_g729-ast14-gcc4-glibc-pentium3.so ' failed.
[Dec 22 15:52:45] WARNING[4491]: loader.c:757 inspect_module: Module
Hi Dave,
context=openbts
callerid=473520
I see you are using OpenBTS. To my understanding, OpenBTS does not
support caller ID, so I don't think it can work.
But as I have the same issue as you, I'd be glad to be wrong ! :D Let me know.
Disregard my answer. I just tested the callerid on my
I'm going to guess you aren't going to get a lot of help on a list
hosted by Digium on how to use a potentially illegal codec...
That said, ast14 in the filename might signify what the problem is.
The APIs likely changed for modules between 1.4 and 1.8.
On Wed, Dec 22, 2010 at 7:58 AM, Giorgio
I don't think a module compiled for Asterisk 1.4 will work with any
other Asterisk version.
On Wed, Dec 22, 2010 at 8:58 AM, Giorgio Incantalupo
gincantal...@fgasoftware.com wrote:
pbx18*CLI module load codec_g729-ast14-gcc4-glibc-pentium3.so
Unable to load module
I see the same thing. Why is there an CANCEL status if it is never set. The
only way I have been able to capture a Cancel status is with the
h extensions using the 'e' option under dial. But this leaves no way to
tell what the DIALSTATUS state was as it is blank. I belive it is a bug as
well.
To my knowledge there is currently no free version of the g729 codec. There
were some spec builds but those were just for testing if I recall
correctly. Each version of the codec that we have always gotten has been
compiled for each version of asterisk. I would just buy the Digium licenses
Hi list,
I have searched through asterisk command lines but haven't found how to do this:
- can I list the phones (callerid or IMSIs?) currently registered ?
If I do dialplan show that lists the configuration I loaded, e.g
[ Context 'sip-local' created by 'pbx_config' ]
'2102' = 1.
Try to use h extension
--
Vardan Harutyunyan,
Senior System Administrator
Enterprise Incubator Foundation
123 Hovsep Emin Street,
Yerevan 0051, Republic of Armenia
Tel: + 374 10 219735
Fax: + 374 10 219777
E-mail: i...@eif.am
www.eif-it.com
Michael wrote:
Hi Nikhil,
Both debug and verbose
Hi all,
thanks for answering.
You all are right but I do not really need the codec because my phones
and my Voip lines are all working using g729. Asterisk is working fine
without transcoding as well.the problem is my CLI is flooded with
messages like:
WARNING[7831] translate.c: No
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Axelle
Sent: Wednesday, December 22, 2010 9:15 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] How to list used extensions + assign extension toa
Hi MrHanMan,
you are right...and the mistake is so stupid I've already solved
itwhat a slip! :)
This means I really need a long relaxing period on some exotic
island...or in some cold prison since I'm using an illegal codec!!! :)
Still I do not believe why Asterisk had not complained for a
The Dial Status is not set when accessing it from the h extension.
Bryant
From: Vardan Harutyunyan hvarda...@gmail.com
Sent: Wednesday, December 22, 2010 10:39 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] DIALSTATUS on CANCEL
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Giorgio
Incantalupo
Sent: Wednesday, December 22, 2010 9:49 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] * 1.8:
45K ?
With 45K I can barely pay for gas, tolls, and breakfast. If you guys are such a
fast growing company, probably you can pay better salaries.
CS
On December 22, 2010 at 9:23 AM Jess Hart j...@langleyjames.net wrote:
Job Description: Asterisk MySQL Support Engineer
Fast Growing
Giorgio
You could buy just a couple of licenses 3 to 5. It would get rid of the
messages for the most part and it would give you the ability to transcode
for voicemails and other items requiring transcode. The reason you are
likely getting the messages is there is some kind of transcode
On Wed, Dec 22, 2010 at 9:48 AM, Giorgio Incantalupo
gincantal...@fgasoftware.com wrote:
Hi all,
thanks for answering.
You all are right but I do not really need the codec because my phones and
my Voip lines are all working using g729. Asterisk is working fine without
transcoding as
Hi,
On Wed, Dec 22, 2010 at 9:49 AM, Alex Saavedra
a...@masterline-logistics.com wrote:
I have noticed thar our dahdi-channels.conf has some repeating directives,
for instance for channel 2 (FXO) we have these settings:
;;; line=2 WCTDM/0/1 FXSKS
signalling=fxs_ks
callerid=asreceived
Hi Stephen,
Jeroen, I'm trying to avoid rewriting the outgoing block for the
patterns mentioned above. I've placed a pseudo dial-plan below. The
plan needs to dial the 1 and/or also the area code depending on the
pattern they enter. Any tips, thanks.
I find a diaplan much easier to read if
On Wed, 22 Dec 2010 14:31:32 +0100, Stefan Schmidt s...@sil.at wrote:
you have a typicall nat issue. Asterisk receives messages from the phone
but cannot send any messages back (thats why it tries to resend the 200
ok message 6 times).
try setting qualify=yes to your sip peers config to keep the
On Wed, 22 Dec 2010 13:18:38 +, Steve Davies davies...@gmail.com
wrote:
Look in the XLite advanced network settings and disable the 2 timeout
settings (RTP and RTCP?). This is not always necessary, but there are
sufficient cases where the packets XLite expects appear early on, but
do not
Ok I can't get my CDR values to set from the h extension in either 1.6.2 or
1.8 What is wrong? Here is what I found in the cdr.conf
; Normally, CDR's are not closed out until after all extensions are
finished
; executing. By enabling this option, the CDR will be ended before
executing
; the
On Wednesday 22 December 2010 08:23:19 MrHanMan wrote:
On Wed, Dec 22, 2010 at 6:41 AM, Steve Davies davies...@gmail.com
wrote:
On 21 December 2010 22:06, Tilghman Lesher tilgh...@meg.abyt.es
wrote:
On Monday 20 December 2010 14:39:36 Ernie Dunbar wrote:
We have an issue with our Asterisk
What would it do if you
exten = h,1,ResetCDR(w)
exten = h,2,NoCDR()
exten = h,3,DEADAGI(get-unqiueid.php)
I have not tried it but in theory it should write the first CDR and then
kill the write of the second NO ANSWER CDR.
Let me know if it works for you as I may need to do it on some of my h
On Wed, 2010-12-22 at 11:23 -0500, C. Savinovich wrote:
45K ?
With 45K I can barely pay for gas, tolls, and breakfast. If you guys
are such a fast growing company, probably you can pay better salaries.
CS
And you have to know Kalamino! :)
j
On December 22, 2010 at 9:23 AM
On Tue, Dec 21, 2010 at 6:59 PM, Stephen Reese rsre...@gmail.com wrote:
On Tue, Dec 21, 2010 at 7:58 PM, Stephen Reese rsre...@gmail.com wrote:
Is there a way to include:
_NXXNXX
_NXX
_011.
_911
into my current plan:
Sorry, here's the rest.
exten =
2010/12/22 Bryant Zimmerman brya...@zktech.com
Giorgio
You could buy just a couple of licenses 3 to 5. It would get rid of the
messages for the most part and it would give you the ability to transcode
for voicemails and other items requiring transcode. The reason you are
likely getting the
Hi,
Any recent experience to share when using OpenStage phones in SIP mode and
Asterisk ?
What about provisionning (and localization) ?
BLF ?
Audio quality ?
User acceptance ?
Cheers
--
_
-- Bandwidth and Colocation Provided by
This is a NAT issue like noted before.
Try:
localnet=192.168.0.0/ http://192.168.0.0/24255.255.255.0
instead of:
localnet=192.168.0.0/24
http://192.168.0.0/24Also, make sure you have all your VPN connections as
localnet and other side subnet as localnet as well if you are using VPN.
Otherwise,
45K GBP would probably cover breakfast in South London. It's about 70 USD.
--Don
Don Kelly
PCF Corp
People Come First
651 842-1000
888 Don Kell(y)
651 842-1001 fax
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of C.
On Wed, 2010-12-22 at 12:42 -0500, Bryant Zimmerman wrote:
Ok I can't get my CDR values to set from the h extension in either
1.6.2 or 1.8 What is wrong? Here is what I found in the cdr.conf
; Normally, CDR's are not closed out until after all extensions are
finished
; executing. By
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bryant
Zimmerman
Sent: Wednesday, December 22, 2010 11:43 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc: Asterisk Developers Mailing List
Subject: Re:
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff
LaCoursiere
Sent: Wednesday, December 22, 2010 11:56 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Vacancy -
On Wednesday 22 December 2010 11:42:33 Bryant Zimmerman wrote:
Ok I can't get my CDR values to set from the h extension in either 1.6.2
or 1.8 What is wrong? Here is what I found in the cdr.conf
; Normally, CDR's are not closed out until after all extensions are
finished
; executing. By
My understanding is that you need one license for every channel it's
being used on, regardless of whether the server is physical or
virtual.
On Wed, Dec 22, 2010 at 12:20 PM, Olivier oza_4...@yahoo.fr wrote:
2010/12/22 Bryant Zimmerman brya...@zktech.com
Giorgio
You could buy just a couple
On Wednesday 22 December 2010 12:20:36 Olivier wrote:
2010/12/22 Bryant Zimmerman brya...@zktech.com
Giorgio
You could buy just a couple of licenses 3 to 5. It would get rid of
the messages for the most part and it would give you the ability to
transcode for voicemails and other
C. Savinovich wrote:
45K ?
With 45K I can barely pay for gas, tolls, and breakfast. If you guys
are such a fast growing company, probably you can pay better salaries.
CS
Isn't that in UK money?
Or Euros?
JN
On December 22, 2010 at 9:23 AM Jess Hart j...@langleyjames.net wrote:
Wouldn't that be 70K USD? Or should we REALLY be worried about the British
economy?
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Don Kelly
Sent: Wednesday, December 22, 2010 12:24 PM
To: 'Asterisk Users Mailing List -
Can you point out to me the places in London that sell food at American prices?
Perhaps I get SeamlessWeb to deliver every morning from Brooklyn to London.
On December 22, 2010 at 1:24 PM Don Kelly d...@donkelly.biz wrote:
45K GBP would probably cover breakfast in South London. It's about
Hi!
Any recent experience to share when using OpenStage phones in SIP mode
and Asterisk ?
I found these phones not to be very comfortable to use, even though they do
look interesting and the hardware is well done. If I remember well you can run
your on JAVA
(and/or XML) applications on it
Gerald,
Thank you for the explanation. Glad I asked...
Alex Saavedra
On Wed, Dec 22, 2010 at 12:40 PM, Gerald A geraldabli...@gmail.com wrote:
Hi,
On Wed, Dec 22, 2010 at 9:49 AM, Alex Saavedra
a...@masterline-logistics.com wrote:
I have noticed thar our dahdi-channels.conf has some
Wait, is 70k US for an experienced engineer supposed to be adequate?
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny
Nicholas
Sent: Wednesday, December 22, 2010 2:27 PM
To:
Asterisk Version: 1.8.1.1
Problem: Multiple Parking Lots
Issue: Not redirecting to the right parking lot. Always uses the first
parking lot from parkedcalls show or features show
Asterisk Working Version: 1.6.1
Steps Taken:
In features.conf added:
[parkinglot_test]
context = parkedcalls-test
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Watkins,
Bradley
Sent: Wednesday, December 22, 2010 2:49 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Vacancy - Asterisk MySQL
Well, I downgraded this box to Asterisk 1.4.38 and all is well again. Echo
cancellation works properly, no problems, no errors.
I have to assume this is a bug in Asterisk 1.8.x or Wanpipe 3.5.18.
--Tim
- Original Message -
Greetings folks-
I'm experiencing issues with a freshly
Hi Mike,
Fork will generated 2 CDRs, and will seperate CDRs
But seems that there is a trouble in 1.6 (1.4 was working fine)
For exemple : phone A (leg A) is called, I play some background sound and
before putting in relation with phone B (leg B) I do a ForkCDR()
in 1.4 billsec in the first
Wait, is 70k US for an experienced engineer supposed to be adequate?
Thank you, not only that , but also note that it would be 70K at the US dollar
exchange rate. However, because it is 45K Euros/Pounds earned and spent in UK,
for all practical purposes it is just the same as if it was 45K US
By UK standards that's a pretty good salary.
Bear in mind that there is no real 1:1 parity in IT salaries. In the US
we earn significantly more for our IT efforts than in the UK.
To give you an example, when I moved from London to New York I got a 4
fold pay rise in real terms for doing
Lots of unemployed engineers in the US would be more than happy with
70K, or even less.
A long period of high unemployment in the US, and world markets is
something many have yet to come to understand.
John Novack
Watkins, Bradley wrote:
Wait, is 70k US for an experienced engineer supposed
have searched this list and others, and see other pepole having this issue.
However, I have not seen how to fix it.
Sep 12 18:52:36 WARNING [4620]: chan_sip.c:1835 retrans_pkt: Maximum retries
exceeded on transmission 778f89593967725f0abe40eb1752504c (at) 10.10.206.53 for
seqno 1620
My h extension is in the same context as my Dial commands. Here is a cut
back version of the code.
The cause_code value is never stored in the mysql databae even when set in
the h extension or the
when set in rc-ANSWER' OR doDialStd
[macro-OBD-DoOutboundDial]
exten = s,1,Macro(${ARG1})
exten =
No this is just a snip of the much larger code.
The h extension is runing but no values port dial function aer being written.
If I do a Set(CDR(field)=Value) before the dial
The value is stored. See my other response for a larger snip of code.
Bryant
On Wed, Dec 22, 2010 at 12:59 PM, Warren Selby wcse...@selbytech.com wrote:
On Tue, Dec 21, 2010 at 6:59 PM, Stephen Reese rsre...@gmail.com wrote:
On Tue, Dec 21, 2010 at 7:58 PM, Stephen Reese rsre...@gmail.com wrote:
Is there a way to include:
_NXXNXX
_NXX
_011.
_911
On Wed, Dec 22, 2010 at 10:12 PM, Stephen Reese rsre...@gmail.com wrote:
Thanks Warren, that's what I'm looking to do.
First question is where did ${MACRO_EXTEN} come from, I assumed
${EXTEN} is a built in variable?
Secondly, where would the 1 and/or area-code need to be placed? Could
an
On Wednesday 22 December 2010 21:08:56 Bryant Zimmerman wrote:
My h extension is in the same context as my Dial commands. Here is a
cut back version of the code.
The cause_code value is never stored in the mysql databae even when set
in the h extension or the
when set in rc-ANSWER' OR
Hi
I am new to asterisk. I want to build an IVR system so that approximately
10-15 users can call simultaneously and use the same dialplan..
We have PRI lines and are thinking of buying Digium TE!21 card for my
software.Would it serve my needs?
Please let me know how to configure dialpan and other
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Cary Fitch
Sent: Wednesday, December 22, 2010 7:49 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Maximum E1
I have make test in AEL.
context fu {
_000./userN = {
Dial(SIP/${EXTEN:3...@prov);
Noop(${DIALSTATUS});
};
h = {
Noop(${DIALSTATUS});
};
};
And look CLI
-- Executing [00018185402...@fu:1] NoOp(SIP/userN-b6317738, )
in new stack
-- Executing
Aloha,
Is there a way to forward a message to multiple people from within the
telephone user interface? Now there is only the ability to forward to
an individual.
I see there is a way to leave a message for multiple people using the
dial plan but that is not available when you are listening to
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