Hi,
I'm trying to setup Asterisk as an outgoing SIP dial tester. There will
be no phones connected to this installation, and I don't need to
process incoming calls. I just need to dial a number, have the person
acknowledge the call, and log that fact. (Basically an automated soft
phone). I
On 07/08/08 11:05, Matt Riddell wrote:
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Joseph Jacobson wrote:
Hi,
I'm trying to setup Asterisk as an outgoing SIP dial tester. There will
be no phones connected to this installation, and I don't need to
process incoming calls. I just need
On 07/08/08 11:55, Matt Riddell wrote:
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Ok, if you type:
core set verbose 10
and
core set debug 10
Then drop the file into /var/spool/asterisk/outgoing
a) does the file disappear
b) does anything come up in the console
c) what is the date on the
On 07/07/08 21:40, Joseph Jacobson wrote:
On 07/08/08 11:55, Matt Riddell wrote:
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Ok, if you type:
core set verbose 10
and
core set debug 10
Then drop the file into /var/spool/asterisk/outgoing
a) does the file disappear
b) does anything come up
In 2006 users complained already the link was dead
It seems the code enjoys the electronic paradise
Not too good for us
Anybody has a better link ?
(if so, update the wiki too please)
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of EdPimentl
Sent: Wednesday 11
Hi,
Did you contact digium ?
I did contact them some years ago with a similar problem (but different)
and they were helpful.
Good luck
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Guilherme Loch
Waltrick Góes
Sent: Monday 24 March 2008 17:59
To: Asterisk
BEGIN:VCALENDAR
PRODID:-//Microsoft Corporation//Outlook 11.0 MIMEDIR//EN
VERSION:2.0
METHOD:REPLY
BEGIN:VEVENT
DTSTART:20080307T11Z
DTEND:20080307T12Z
LOCATION:http://voipusersconference.org
TRANSP:OPAQUE
SEQUENCE:0
UID:[EMAIL PROTECTED]
DTSTAMP:20080306T094011Z
SUMMARY:Declined: VoIP
is
not\n\nwhat time is it then ? 12-13 Paris time or eastern us ?\n\nbest
regards\n\nt. jacobson\n\n \n\n
SUMMARY:Tentative: VoIP Users Conference
PRIORITY:5
X-MICROSOFT-CDO-IMPORTANCE:1
CLASS:PRIVATE
ATTENDEE;PARTSTAT=TENTATIVE:MAILTO:[EMAIL PROTECTED]
END:VEVENT
END:VCALENDAR
No virus found
changers
good trip
Shaoul Jacobson
Senior VoIP Consultant
Tellink
Tel : +32 3 201 96 36
Fax : +32 3 227 09 81
e-mail [EMAIL PROTECTED]
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Hi,
Try to cancel 'silence suppression' from
the 'other' source.
Asterisk does not support 'silence suppression'
(yet ?) (as far as I know)
Regards,
Shaoul Jacobson
Senior VoIP Consultant
Tellink
Tel : +32 3 201 96 36
Fax : +32 3 227 09 81
e-mail [EMAIL PROTECTED
Hi,
Try WITHOUT silence suppression.
* does not seem to support that
regards,
Shaoul Jacobson
Senior VoIP Consultant
Tellink
Tel : +32 3 201 96 36
Fax : +32 3 227 09 81
e-mail [EMAIL PROTECTED]
-Original Message-
From: Asterisk guy [mailto:[EMAIL PROTECTED]
Sent: mardi 3 mai
will provide similar (or better) explanations
Shaoul Jacobson
Senior VoIP Consultant
Tellink
Tel : +32 3 201 96 36
Fax : +32 3 227 09 81
e-mail [EMAIL PROTECTED]
-Original Message-
From: Mr AG!! [mailto:[EMAIL PROTECTED]
Sent: lundi 2 mai 2005 11:12
To: asterisk-users@lists.digium.com
a switch (gives power)
Plugging 2 FXS elements together is not a good idea
Plugging 2 FXO elements together will not work but no problem otherwise
Shaoul Jacobson
Senior VoIP Consultant
Tellink
Tel : +32 3 201 96 36
Fax : +32 3 227 09 81
e-mail [EMAIL PROTECTED
Hi,
What do you have h323 or oh323 ,
(open h 323)
I think you have the latest.
You must use the SPECIFIC files.
Check http://www.inaccessnetworks.com/projects/asterisk-oh323
Also PATCH the file BEFORE compilation
It should run then.
Good luck
Regards,
Shaoul Jacobson
Senior VoIP Consultant
Hi,
The legal way is to buy a smartnet (support contract) for the soft.
That way you can download it from Cisco's
web site.
Try to contact your reseller.
Regards,
Shaoul Jacobson
Senior VoIP Consultant
Tellink
Tel : +32 3 201 96 36
Fax : +32 3 227 09 81
e
our system.
Also dbhost=127.0.0.1
Also check mysql
to see if THAT user may connect from THAT machine 127.0.0.1
Good luck
I assumed you use mysql
and connect with mysql socket on localhost
Adapt if you use odbc
or another host
Shaoul Jacobson
Senior VoIP Consultant
Tellink
Tel :
Jacobson
Senior VoIP Consultant
Tellink
Tel : +32 3 201 96 36
Fax : +32 3 227 09 81
e-mail [EMAIL PROTECTED]
-Original Message-
From: Nathaniel Angelo A. Torres
(247talk) [mailto:[EMAIL PROTECTED]
Sent: mardi 19 avril 2005 10:52
To: Asterisk Users Mailing List -
Non-Commercial
the bugtracker
good luck
Shaoul Jacobson
Senior VoIP Consultant
Tellink
Tel : +32 3 201 96 36
Fax : +32 3 227 09 81
e-mail [EMAIL PROTECTED]
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too.
That sould free some irq.
Also irq 15 (ide1) could be free if you do not put a cd on the second
channel
After freeing irq's, it might be necessary to physically put the cards in
other slots (or swap those if you have too few slots)
Regards,
Shaoul Jacobson
Senior VoIP Consultant
Tellink
Tel
Dear Matthew, thanks a lot.
That did help.
This could be a nice addition to the samples on the wiki.
Regards,
Shaoul Jacobson
Senior VoIP Consultant
Tellink
Tel : +32 3 201 96 36
Fax : +32 3 227 09 81
e-mail [EMAIL PROTECTED]
-Original Message-
From: Matthew Boehm [mailto:[EMAIL
,
Shaoul Jacobson
Senior VoIP Consultant
Tellink
Tel : +32 3 201 96 36
Fax : +32 3 227 09 81
e-mail [EMAIL PROTECTED]
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to another on MPLS, you lose all the
benefits. No control.
Using the World Wide Wait (Internet) it will not help.
A waste of money.
My 2 cents.
Shaoul Jacobson
Senior VoIP Consultant
Tellink
Tel : +32 3 201 96 36
Fax : +32 3 227 09 81
e-mail [EMAIL PROTECTED
not at the backbone.
Regards,
Shaoul Jacobson
Senior VoIP Consultant
Tellink
Tel : +32 3 201 96 36
Fax : +32 3 227 09 81
e-mail [EMAIL PROTECTED]
PS
I have worked in close relations with some 'big' providers.
They accept sla's, backup circuits even when they know they cannot provide
,tr)
I already found out that the commas need to be replaced by '|'.
(exten = ... Dial(SIP/1007,20,tr) becomes ..., 'Dial', '1007|20|tr' )
It is mentioned only for the 'goto' in the wiki.
Maybe is it worth to broaden up the sample.
Regards,
Shaoul Jacobson
Senior VoIP Consultant
Tellink
Tel
this in the sample to make it more clearly for
others.
I still do not know how to 'translate' (from old extension.conf) :
- the '_9.' (playing with variable filters)
- the 'SIP/${EXTEN:1}' (playing with functions and number manip)
regards
Shaoul Jacobson
started.
No sql problems to be seen in log file
Realtime mysql status shows :
connected to [EMAIL PROTECTED], port 3306 wih username asterisk for xx
minutes
so ?
regards,
Shaoul Jacobson
VoIP Consultant
Tellink
Tel : +32 3 201 96 36
Fax : +32 3 227 09 81
e-mail [EMAIL PROTECTED
sip.conf to sip.old
Asterisk -vvc shows realtime has started.
No sql problems to be seen in log file
Realtime mysql status shows :
connected to [EMAIL PROTECTED], port 3306 wih username asterisk for xx
minutes
so ?
regards,
Shaoul Jacobson
VoIP Consultant
Tellink
Tel : +32 3 201 96 36
and from memory
Regards,
Shaoul Jacobson
Senior VoIP Consultant
Tellink
Tel : +32 3 201 96 36
Fax : +32 3 227 09 81
e-mail [EMAIL PROTECTED]
-Original Message-
From: Steve Blair [mailto:[EMAIL PROTECTED]
Sent: mercredi 16 mars 2005 16:35
To: Asterisk Users Mailing List - Non
Hi,
Welcome.
Read the samples *.conf files
(in /etc/asterisk)
extension.conf, sip.conf are
some good places to start.
Read search the wiki.
Many info there (also not always very clear)
success
Shaoul Jacobson
Senior VoIP Consultant
Tellink
Tel : +32 3 201
config.
Please give me a feed-back if those files helped you and how.
Also if you have a work-around (like an old file to use)
Thanks regards,
Shaoul Jacobson
Senior VoIP Consultant
Tellink
Tel : +32 3 201 96 36
Fax : +32 3 227 09 81
e-mail [EMAIL PROTECTED]
-Original Message-
From
recently.
I did today a cvs again.
but the 'missing file' did not came.
And I could not find it in another directory.
Regards,
Shaoul Jacobson
Senior VoIP Consultant
Tellink
Tel : +32 3 201 96 36
Fax : +32 3 227 09 81
e-mail [EMAIL PROTECTED
Jacobson
Senior VoIP Consultant
Tellink
Tel : +32 3 201 96 36
Fax : +32 3 227 09 81
e-mail shaoul (at) tellink.com
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If I remember correctly, Cisco parse the codecs according to their entry
onder. Asterisk orders according to alphabetical order.
If you do not need both codecs, set only one to simplify.
Regards,
Shaoul Jacobson
Senior VoIP Consultant
Tellink
Tel : +32 3 201 96 36
Fax : +32 3 227 09 81
e
Jacobson
VoIP Consultant
Tellink
Tel : +32 3 201 96 36
Fax : +32 3 227 09 81
e-mail [EMAIL PROTECTED]
-Original Message-
From: Sergey Kuznetsov [mailto:[EMAIL PROTECTED]
Sent: jeudi 10 février 2005 4:15
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Please share
Jacobson
VoIP Consultant
Tellink
Tel : +32 3 201 96 36
Fax : +32 3 227 09 81
e-mail [EMAIL PROTECTED]
-Original Message-
From: Patrick[EMAIL PROTECTED]
Sent: 10/02/05 07:28:25
To: Asterisk Users Mailing List - Non-Commercial
Discussionasterisk-users@lists.digium.com
Subject
Jacobson
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, and port
5062 for 555-1214 ? This way each NetMeeting, or GnomeMeeting
connection coming from India can simply run behind a NAT router,
instead of setting up a separate Asterisk PBX.
Thank you for your help.
Cameron Jacobson
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What kind of stability / reliability are people currently experiencing
with the Linux / Asterisk combination? We will be running 3-10 SIP
phones from India to US using nothing more than regular cable / dsl
connections from both locations.
Also, what make / model SIP phone do you recommended
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