Hello ppl,
Searched all over, but couldn't find anything conclusive.
Does an off-the-shelf version of Asterisk run without any issues on a
64-bit machine?
Does anyone have any 'conclusive' figures?
Apologies if this is a repeat question. Would appreciate if I could be
redirected to the
Hello good ppl,
A couple of questions for multiple pbxes
1. Is it possible to support multiple pbxes in one Asterisk box(using
contexts, etc.)?
2. Can we use the domain field in sip.conf to specify the different
domains for sip users, having one domain for each pbx?
I just tried registering two
anyone , to my query(abt multiple pbxes)? Apologies if I
am missing something elementary here.
cheerz
- Ben.
C F wrote:
Can you please get rid of your awfull long nonsense disclaimer?
On 8/1/07, Benjamin Jacob [EMAIL PROTECTED] wrote:
Hello good ppl,
A couple of questions for multiple pbxes
Anthony Francis wrote:
Hello good ppl,
A couple of questions for multiple pbxes
1. Is it possible to support multiple pbxes in one Asterisk box(using
contexts, etc.)?
2. Can we use the domain field in sip.conf to specify the different
domains for sip users, having one domain for each pbx?
I just
Hello All,
Is CALLERID() setting broken in 1.4.4?
My small dialplan :
[testclid]
exten = _0.,1,Set(CALLERID(all)=Ben Jacob 988077)
exten = _0.,n,Dial(SIP/${EXTEN})
Correct me if I am wrong, Set(CALLERID(all) above supposed to change the
display name as above(Ben Jacob) and change the From URI
identities of end subscribers.
All corrections and suggestions welcome.
- Ben
Benjamin Jacob wrote:
Hello All,
Is CALLERID() setting broken in 1.4.4?
My small dialplan :
[testclid]
exten = _0.,1,Set(CALLERID(all)=Ben Jacob 988077)
exten = _0.,n,Dial(SIP/${EXTEN})
Correct me if I am wrong, Set
to confirm
identities of end subscribers.
All corrections and suggestions welcome.
- Ben
Benjamin Jacob wrote:
Hello All,
Is CALLERID() setting broken in 1.4.4?
My small dialplan :
[testclid]
exten = _0.,1,Set(CALLERID(all)=Ben Jacob 988077)
exten = _0.,n,Dial(SIP/${EXTEN})
Correct me if I am wrong
Hello good ppl,
Any way of stopping asterisk writing into syslogs or any other file, if
I set verbose 6 on the console?
All I want is the verbose output only on the console, nowhere else.
My logger.conf says :
console= notice,error
;messages = notice,warning,error
Thanks in advance.
- Benjamin
Btw, even the syslog line in logger.conf is commented :
; syslog.local0 = notice,warning,error
Benjamin Jacob wrote:
Hello good ppl,
Any way of stopping asterisk writing into syslogs or any other file, if
I set verbose 6 on the console?
All I want is the verbose output only on the console
Exactly the same lines as on the console.
Adrian Marsh wrote:
What logs are coming out to /var/log/messages?
Adrian Marsh
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Benjamin
Jacob
Sent: 04 September 2007 07:58
To: Asterisk Users Mailing List - Non
Here it is :
SIP01*CLI logger show channels
Channel Type StatusConfiguration
--- ---
Console Enabled- Notice Error
Tzafrir Cohen wrote:
On Tue, Sep 04,
accessing the * console via ssh.
thanks for ur help.
- Benjamin Jacob.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Benjamin
Jacob
Sent: 04 September 2007 12:51
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] stop
- Benjamin Jacob.
EMAIL DISCLAIMER : This email and any files transmitted with it are
confidential and intended solely for the use of the individual or entity to
whom they are addressed. Any unauthorised distribution or copying is strictly
prohibited. If you receive this transmission in error
C F, I have nat=yes set by default for all my extensions(with
canreinvite=no). And things work fine.
Bilal, about Asterisk sending packets to public/private :
Asterisk will send packets to the public IP advertised by the msg/recv
from address. It is the NAT's headache on the endpoints network
You The Man, Anselm. Thanks for the details.
Anselm Martin Hoffmeister wrote:
Am Dienstag, den 11.09.2007, 17:11 +0530 schrieb Benjamin Jacob:
Thanks Anselm. This does clears a few things for me.
Tho, I couldnt find the patterns you mentioned in the docs(do point me
to the location if you
/debug messages into
/var/log/messages.
Same like your situation, the messages is comment (;)
and even the logges are written to the
/var/log/messages, so why that is happening?
Did u find answer for that?
Regards
Bilal
--- Benjamin Jacob [EMAIL PROTECTED] wrote:
Hello Bilal,
You have to do
safe_asetrisk bundled with the package, does increase the file limits in
quite a neat way, with some other good setups.
Edit MAXFILES or SYSMAXFILES as required.
Also, I've read posts online, advising not to use safe_asterisk. Any
experiences on this one, anyone?
cheers
- Ben.
Jay R. Ashworth
a2billing so far seems to be quite comprehensive compared to the other
freeware asterisk-based billing solutions available out there.
We are building our own billing solution(due to the very peculiar
requirements, one of which is to bill the callee, rather than the
caller). We are achieving
show us the output of ur top command
Pezhman Lali wrote:
Dear
I am using an asterisk 1.2.7.1 , with postgres
and safe_Asterisk, for running, asterisk.
but there is a problem,
after 2-3 hours after restarting any things, top
shows me, that, two asterisk, are now running, and one
of them, gets
Hello Steve,
I think Ray was talking more like the following setup (do correct me if
I am wrong):
User A (SIPcallId1) --- Asterisk (SIPcallId2) -- User B
In this case, the INVITE SIP callId received by Asterisk from User A is
different to that sent in the INVITE to User B.
I can get User
Also, how do you acces the second SIP call ID from the dialplan? Any
simple way to do this?
Benjamin Jacob wrote:
Hello Steve,
I think Ray was talking more like the following setup (do correct me
if I am wrong):
User A (SIPcallId1) --- Asterisk (SIPcallId2) -- User B
In this case
Hello All,
I've got the following warning messages a couple of days back:
/chan_sip.c: Maximum retries exceeded on transmission SIPcallId for
seqno 1 (Critical Response).
/Have got the warnings repeatedly for one Callid. If maximum retries
have exceeded why should it give me those warnings
Hello ppl,
Is there any other way to detect states like Ringing on SIP channels on
Asterisk?
Nvlinedetect is one way, but it seems to have disappeared from the face of the
earth!
Any pointers or does anyone have the code for NV* features?
Thanks in advance
- Ben.
Hello ppl,
I am using the Astman API Originate command to initiate a call to a user. On
connect of the user, I dial another user to bridge the call between the two.
I am using the Async option with the Originate command, as I don't want to use
Astman proxy yet. Is there any way to invoke a
Hello ppl,
Any way to do a re-invite and make RTP bypass Asterisk, after call
establishment.
In other words, I would like to control when to do the bypass work for
peer-peer RTP flow.
The issue is that I need to send DTMFs after dialing the user because
most of the users are behind PBXes
Hello ppl,
Any way to do a re-invite and make RTP bypass Asterisk, after call
establishment.
In other words, I would like to control when to do the bypass work for
peer-peer RTP flow.
The issue is that I need to send DTMFs after dialing the user because most of
the users are behind PBXes
clear this time.
cheerz
- Ben.
Steve Davies [EMAIL PROTECTED] wrote: On 21/04/2008, Benjamin Jacob wrote:
Hello ppl,
Any way to do a re-invite and make RTP bypass Asterisk, after call
establishment.
In other words, I would like to control when to do the bypass work for
peer-peer RTP flow
bridge and I need to send DTMF (the
bridge PIN) to it after connection. But alas, the reinvite happens before the
D() is executed.
The SIP gateway is MySIPGateway at 204.aaa.bbb.ccc.
cheers
- Ben.
Steve Davies [EMAIL PROTECTED] wrote: 2008/4/22 Benjamin Jacob :
[snip]
So, my question : once
Hello ppl,
One on my clients' machine had Asterisk 1.4.4. installed. The complained of
choppy Playback of gsm files.
So scouring the internet gave me the solution of installing ztdummy and loading
it as a module.
Did it (using zaptel-1.4.1) , but to no effect. Re-compiled asterisk and
Are my messages getting through?
This is urgent!! Any pointers?
Benjamin Jacob [EMAIL PROTECTED] wrote: Date: Thu, 24 Apr 2008 23:23:08 -0700
(PDT)
From: Benjamin Jacob [EMAIL PROTECTED]
Subject: Playback / Background / Read choppy, but musiconhold fine, even with
ztdummy
To: asterisk-users
Tony Mountifield [EMAIL PROTECTED] wrote: In article [EMAIL PROTECTED],
Benjamin Jacob wrote:
One on my clients' machine had Asterisk 1.4.4. installed. The complained of
choppy Playback
of gsm files.
So scouring the internet gave me the solution of installing ztdummy and
loading
Benjamin Jacob [EMAIL PROTECTED] wrote:
Tony Mountifield [EMAIL PROTECTED] wrote: In article [EMAIL PROTECTED],
Benjamin Jacob wrote:
One on my clients' machine had Asterisk 1.4.4. installed. The complained of
choppy Playback
of gsm files.
So scouring the internet gave me the solution
OK, I think you need to home in on the differences
between the server(s)
that work fine and the one that doesn't.
As I said in my other mail, the faulty one is a
.. mono processor machine, with SMP turned on
.. running CentOS 5
.. with kernel : 2.6.18-53.1.13.el5
There are other kernels
http://www.openvox.com.cn/products_detail.php?genre_id=9id=28
If you can get the bare card, you can use it for
timing with a little
magic that can be found via google. If not, get one
with an FXO or
FXS and you will add a little flexibility and have
real hardware
timing.
If you
-
In the process of cleaning up unnecesary
processes, I
came across this line :
/usr/sbin/vmware-guestd --background
/var/run/vmware-guestd.pid
GASP so does this mean this is a virtual
machine??
I have got no idea about virtualization yet. So
how do
I confirm if
Hiren,
Not really clear as to what are the things you exactly
want.
List them out clearly.
Before you do that, do google and read up on
Asterisk+IVR
Asterisk+agi
Your need to calculate sum of birthdate digits etc can
be achieved using AGI scripting.
cheers
- Ben.
--- Hiren Mistry [EMAIL
Last I was working on it, it did indeed NOT look at
sip.conf with realtime architecture being used.
But why take chances anyway? Move all the relevant
conf files from /etc/asterisk to some other place to
be safe.
cheers
- Ben.
--- Rilawich Ango [EMAIL PROTECTED] wrote:
HI,
Does asterisk
Hello ppl,
Anyway in Asterisk to update a DB/ do some action on
events like ringing.
The issue is I need to be able to hangup/cancel a
call, if it's ringing(decided by the admin). This is
independant of the timeout that we can specify in the
Dial command.
If I could somehow update a DB with
Hello ppl,
Are the channel names generated on 'Dial's supposed to be unique?
I see the channel names repeating on my asterisk box. I just wanted to confirm
this.
Can anyone point me to the lines of code where the channel name is
generated/calculated? I tried looking, but it looks like quite a
, May 15, 2008, 11:46 AM
Benjamin Jacob wrote:
Are the channel names generated on 'Dial's
supposed to be unique?
I see the channel names repeating on my asterisk box.
I just wanted to confirm this.
Can anyone point me to the lines of code where the
channel name is generated/calculated? I
ringing event
To: asterisk-users@lists.digium.com
Date: Thursday, May 8, 2008, 12:00 AM
On Thu, May 08, 2008 at 12:19:52AM +0300, Atis Lezdins
wrote:
On Wed, May 7, 2008 at 5:43 PM, Philipp Kempgen
[EMAIL PROTECTED] wrote:
Benjamin Jacob schrieb:
Anyway in Asterisk to update a DB
Hello ppl,
Is there anyway to control a call mid-way in terms of sending a re-INVITE with
say sendonly, etc. to mute one call leg of a bridged call ??
Looked around, so far, doesnt seem to be possible.
If it's not, I think it's quite an important feature (re-INVITES mid-session)
for a B2BUA.
I think, you can use the 'h' extension to invoke scripts (DeadAGI to be more
precise) on hungup channels.
use something like this :
exten = _X., 1, NoOp(got a call)
exten = _X., n, Dial(somexten}
exten = h, 1, DeadAGI(hangupScript.sh)
--- On Fri, 6/13/08, Robor Oghene [EMAIL PROTECTED]
Care to explain the scenario Vincent?
Is it a SIP peer?
what is the DTMF mode set? etc.
--- On Tue, 7/1/08, Vincent [EMAIL PROTECTED] wrote:
From: Vincent [EMAIL PROTECTED]
Subject: [asterisk-users] User unable to use DTMFs?
To: asterisk-users@lists.digium.com
Date: Tuesday, July 1,
If by realtime, you mean to be able to read the MOH class from a DB and set
MusicOnHold, then I think you should try func_odbc.
Have never tried it, but reading the workings of it, it seems to be possible to
achieve this.
Let me know if you succeed in it.
- Ben.
--- On Tue, 7/1/08, Nhadie
modprobe zaptel; modprobe ztdummy
That will start zaptel and ztdummy after the 'zaptel stop'. Then restart
asterisk.
--- On Wed, 7/2/08, Doug Crompton [EMAIL PROTECTED] wrote:
From: Doug Crompton [EMAIL PROTECTED]
Subject: Re: [asterisk-users] Choppy audio
To: Asterisk Users Mailing
Use SendDTMF.
--- On Thu, 7/3/08, Neha Punia [EMAIL PROTECTED] wrote:
From: Neha Punia [EMAIL PROTECTED]
Subject: [asterisk-users] (no subject)
To: asterisk-users@lists.digium.com asterisk-users@lists.digium.com
Date: Thursday, July 3, 2008, 10:29 AM
Hi
I m making a call from one
Hello Roland,
You can use the cmd Read for this.
http://www.voip-info.org/wiki/view/Asterisk+cmd+Read
Pretty straight forward. Whenever you need to accept DTMF input from the user
collect the required digits using Read; check the collected digits; if yes jump
to required extension; else
Look at the canreinvite option.
- Original Message
From: Rizwan Hisham [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, September 17, 2008 3:20:40 PM
Subject: [asterisk-users] dtmf passthru
hi all,
Is
From: Benjamin Jacob [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, September 17, 2008 4:45:13 PM
Subject: Re: [asterisk-users] dtmf passthru
Look at the canreinvite option.
- Original Message
From
Hi Bilal,
Yes it is definitely possible. And I've done it myself for a couple of our
clients.
Does that answer your two questions?
cheers
- Ben.
--- On Tue, 9/23/08, bilal ghayyad [EMAIL PROTECTED] wrote:
From: bilal ghayyad [EMAIL PROTECTED]
Subject: [asterisk-users] AGI and prepaid
If you already havent seen this:
http://dir.indiamart.com/impcat/video-telephone.html
cheerz
- Ben.
Crazy Boy wrote:
Hi Friends,
This is Chandra from India. I have installed and configured Asterisk
in our company. I want to provide Polycom IP 501 model phones to our
employees. I am unable
Make it
Goto(s-${DIALSTATUS})
cheerz
- Ben.
Yuan LIU wrote:
In examples, s-${DIALSTATUS} is used to handle unsuccessful dial
attempts in the s extension. Goto() is used in examples. Is the
prefix s- mandatory? Is it related to the original extension s?
(Apparently Goto(${DIALSTATUS})
Pierre,
Thats exactly what Joanna said in her reply.
Check the client DTMF settings on your phones.
set it to rfc2833 or out-of-band, whatever the config says.
Grandstream by default have inband DTMF set, and usualy ulaw is
supported as well, and thats the reason ur grandstream works but
rfc2833 is the prefered way, as inband will work perfectly only with the
ulaw codec.
Pierre Marceau wrote:
Okay, in the SPA-941 admin I changed:
;DTMF Tx Method: Auto
DTMF Tx Method: Inband
and now it works.
Thanks!
Pierre
[EMAIL PROTECTED] 2/21/2007 12:09 AM
Pierre,
Thats
Have canreinvite set for your internal extens.
You can also have canreinvite enabled by default for all and use one or more of
the 't','T','h','H','w','W' or 'L' options set in your dial commands which will
override the canreinvite option and not send re-invites.
cheers
- Ben
--- On Sat,
Hello All,
Am relatively new to Asterisk, but kinda slogging my ass off on it.
My first couple of qs to begin with :
1) I tried the voicemail on no-answer thing. and my line in the
voicemail.conf, duz have an email address and also attach=yes,
5600 = 5600, Benjamin Jacob, [EMAIL PROTECTED
Thanks for the sendmail tip guys.
Now the 2nd q was the more urgent one and still is.
How on earth do you edit cofigurations in Asterisk. (na.. am not talking
thru your fav editor).
Like say a web application wants to add an exten, or change the
forwarding of some extension, etc. all this
Ravi,
Have you made the entry for 9001 in voicemail.conf as well??
this entry will be the mailbox config, where you can specify the
password, email add, etc.
- Benjamin Jacob.
raviprakash sunkara wrote:
Hi Users,
I'm new to Asterisk, and I'm working with openSER ,
For Call Routing I'm
voicemail.conf) ,
5600 = 5600, Benjamin Jacob, [EMAIL PROTECTED]|attach=yes
In maillog, I see the To address to be [EMAIL PROTECTED]|attach=yes .
Is it some bug in asterisk mailing code, or am i doing something wrong?
Ben.
Tim St. Pierre wrote
Hello ppl,
Is it possible to include contexts in the RealTime scenario??
If not, wots the work around??
Thanks in advance.
Ben.
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options
Rushowr wrote:
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Benjamin Jacob
Sent: Monday, September 04, 2006 8:37 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] includes in realtime ??
Hello ppl
Exactly my point!
In my earlier mail, I had a typo in my command. I meant n again tried
the command
realtime load sipusers name 4000
and also
realtime load sipusers username 4000
Its not working yet!
Also, if Realtime, I shudn't even be having the need to use the
realtime load commands!!
:) , done all!!
neway, lemme know if am overlooking something.
extconfig.conf
==
sipusers = mysql,astDb,sip_conf
sippeers = mysql,astDb,sip_conf
voicemail = mysql,astDb,voicemail_conf
extensions = mysql,astDb,extensions_conf
sip.conf has got all entries commented, except for
[general]
If it shows in the show modules command, it means, the module is loaded,
right?
If yes,
^CLIshow modules like app_re
Module Description
Use Count
app_realtime.soRealtime Data Lookup/Rewrite 0
app_readfile.so
Hello all,
Another question related to Realtime.
Is it possible to call macros using Realtime arch?
I have a macro definition in table extensions_conf in my MySql db as:
30 | macro-stdpbx1exten | s |
1 | SET|
Hello ppl,
Ive a couple of macros defined to call fwd based on time to a
number/voicemail.
Very elementary.
=
11. [macro-dialexten]
12. exten = s,1,Dial(SIP/${ARG1}) ;
1. [macro-stdpbx1exten]
2. exten = s,1,Set(fwdedNum=${DB(CFWD/${ARG1})})
3. exten =
Thanks Simon,
will try and get back on this one .
Ben.
Simon Woodhead wrote:
Hi Ben,
Yes it is but you need to remember to still include
[macro-stdpbx1exten]
switch = Realtime/
in your extensions.conf
Simon
On 9/6/06, * Benjamin Jacob* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote
lol RR.
will def do some RnD on this one, and wil get back. have put this on the
back burner for now.
thanks again.
cheerz
Ben
RR wrote:
I use rtcachefriends=yes and any changes I make in my database become
effective immediately along with also getting the MWI functionality.
Even though
Hello ppl,
Ive a couple of macros defined to call fwd based on time to a
number/voicemail.
Very elementary.
=
11. [macro-dialexten]
12. exten = s,1,Dial(SIP/${ARG1}) ;
1. [macro-stdpbx1exten]
2. exten = s,1,Set(fwdedNum=${DB(CFWD/${ARG1})})
3. exten =
-0500
Single quotes - ' - work when I set other variables that contain special
characters. Give that a try,
-Tim
On September 6, 2006 23:18, Benjamin Jacob wrote:
Hello ppl,
Ive a couple of macros defined to call
)
==
Basicaly, am storing individual entries of time, and putting them back
together in GotoIftime.
If anyone's got a better solution, lemme know.
cheerz.
Ben.
Benjamin Jacob wrote:
Nope Tim,
had tried that already, duznt work.
Here's the cli output
===
Executing Set(SIP/4000
My q too!!
I mean, simple extension numbers, 3000, 3001, etc, can be present in
multiple pbxes(in any hosted pbx service).
I guess 'context' is a way, but it seems, in *, the dialplan(and hence
the context) is decided by the callee digits. Any work arnd over this one?
cheerz
Ben.
Ricardo
Hello ppl,
Wanted to know your experiences, if you've worked with Asterisk Realtime
Architecture.
Which one do you prefer, static or realtime?
I personaly think, the static architecture is a better solution, cuz, in
the realtime config, to check the dialplan(n hence the sql database) for
Rushowr wrote:
Benjamin Jacob wrote:
Hello ppl,
Wanted to know your experiences, if you've worked with Asterisk Realtime
Architecture.
Which one do you prefer, static or realtime?
I personaly think, the static architecture is a better solution, cuz, in
the realtime config, to check
Hello ppl,
Any idea how do I write in include lines(for contexts or include files)
in the database, in the ARA static config?
thanks in advance.
Ben.
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To
Rushowr wrote:
Hugo wrote:
Anyone could help to use Static RealTime with SIP.CONF. I use Dynamic
Realtime successfully. In fact, I want to know how to compos the correct
DB(postgres or mysql) fields (I think STATIC configuration is different
from DYNAMIC).
Regards,
Hugo
, but it
didn't work. No users have been found (sip show conf).
If you could help me to solve my problem, I would be tkankful
regards
2006/9/12, Benjamin Jacob [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED]:
Rushowr wrote:
Hugo wrote:
Anyone could help to use Static RealTime
.
cheerz
Ben.
Benjamin Jacob wrote:
Hello ppl,
Any idea how do I write in include lines(for contexts or include
files) in the database, in the ARA static config?
thanks in advance.
Ben.
___
--Bandwidth and Colocation provided by Easynews.com
Hello ppl,
I am getting the following errors when accessing voicemails
Sep 13 16:43:59 ERROR[19020]: app.c:1161 ast_lock_path: Unable to create
lock file '/var/spool/asterisk/voicemail/pbx1VmBoxes/555123/Old': No
such file or directory
Sep 13 16:43:59 ERROR[19020]: app.c:1196 ast_unlock_path:
Hello ppl,
Am trying to build a system, wherein users can access their profiles,
and hence voicemails thru a browser.
I am using Apache and am running it on another box and asterisk on
another. Am keeping them seperate to not have http traffic on the same
box as asterisk.
Now, my qs:
Is
THanks RR,
Am trying it right now.
But getting into all kinds of trouble!! ranging from SQL Alloc failed -
to seg faults.
I honestly donno anything abt odbc(n i seriously dont have the time to
rnd on that one, right now)
Can you or anyone else paste their config files for(related to voicemail
, how on earth do i read the recordings and play them out thru a
browser!!
pheww. it never ends!!!
Benjamin Jacob wrote:
THanks RR,
Am trying it right now.
But getting into all kinds of trouble!! ranging from SQL Alloc failed
- to seg faults.
I honestly donno anything abt odbc(n i seriously
Rushowr wrote:
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Rushowr wrote:
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Kristian Kielhofner wrote:
Benjamin Jacob wrote:
Rushowr wrote:
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk
Hello ppl,
I had appdata set to use the function ENUMLOOKUP. But it gets me nothing.
| id| context | exten | priority | app |
appdata
, but the function ENUMLOOKUP doesn't(it just
searches for e164.arpa, and if not found, gives up, if the zone argument
is left empty).
Anyone worked around this one??
cheerz
Ben.
Benjamin Jacob wrote:
Hello ppl,
I had appdata set to use the function ENUMLOOKUP. But it gets me nothing.
| id
Didnt quite get ur question.
But, if you mean, you want to, for e.g. play a file, dial out another
number, sing a song, dance around, after execution of VoicemailMain,
yes, its very much possible. Just add your enhanced dialplan at
the next priority of VoicemailMain.
cheerz
- Ben
Jay R. Ashworth wrote:
On Tue, Sep 26, 2006 at 05:33:15PM -0500, DiegoF wrote:
hola a todos, tengo una duda, ye he resuelto algunas pero otras llegan,
/
hello to all, I have a doubt, ye I have solved some but others arrive,
Hello ppl,
Is it possible to get a count of the number of calls, for which the
media is passing thru Asterisk and for calls, which are bypassing
Asterisk for media?
Thanks in advance.
Ben
___
--Bandwidth and Colocation provided by Easynews.com --
Hello ppl,
follow up on a somewot old post.
I set rtcachefriends=no and voila! changes to codecs, etc are
immediately reflected!
now.. that duz raise some issues .. hmmm
cheerz
Ben.
Douglas Garstang wrote:
If you want to use MWI, and I imagine most people would, you have to cache your
Giorgio Incantalupo wrote:
Hi,
I'm using mysql to store my cdr data. I compiled asterisk-addon module
without problems and I see nothing unusual in my cdr_mysql.conf but
when I do a reload I get this messages (never seen before):
Oct 16 09:43:16 WARNING[8576]: db.c:67 dbinit: Unable to open
Something better would be
http://www.voip-info.org/wiki/index.php?page=Asterisk+tips+openhours
to have includes based on time
cheerz
- Ben
Mohamed A. Gombolaty wrote:
Dear Moj,
Thanks a lot fo the tip, it seems I can do that it is very flexible
and easy to use, I will try to add it to the
Hello ppl,
This post is to do with the variables 'nat' or 'canreinvite' for sip
entities.
Idealy users, wont be static, they could be roaming all over the globe.
So, setting someone as behind NAT, and disabling canreinvite, etc.,
restricts the roaming capabilities of a user.
Is there any way
Hello ppl,
Is it possible to send SIP messages as response to the calling UA on
failure, for e.g. if a number is blacklisted I would want to send
Forbidden to the caller, not just for user comfort but also for testing
purposes?
I see only Congestion available which sends Service Unavailable.
Magnusson wrote:
Benjamin Jacob wrote:
Hello ppl,
Is it possible to send SIP messages as response to the calling UA on
failure, for e.g. if a number is blacklisted I would want to send
Forbidden to the caller, not just for user comfort but also for
testing purposes?
I see only Congestion available
On Tuesday 17 October 2006 10:31, Time Bandit wrote:
The one that never did a mistake, never did anything
so the q is.. will you be doing something a lot?? ;-)
... just kidding mate.. but thats a good line neway.
cheerz
___
--Bandwidth
Mohamed A. Gombolaty wrote:
Dear Rich,
It seems that my question is very general I apologize for that, but I
am glad to see others like yourself pointing me in different
directions, it seems all around the world we have problems with the
cleaning folks.
What I have in mind is to make the
Eric ManxPower Wieling wrote:
Benjamin Jacob wrote:
Hello ppl,
This post is to do with the variables 'nat' or 'canreinvite' for sip
entities.
Idealy users, wont be static, they could be roaming all over the
globe. So, setting someone as behind NAT, and disabling canreinvite,
etc
1 - 100 of 139 matches
Mail list logo
