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Philipp Kempgen wrote:
Tobias Ahlander schrieb:
From: Mark Michelson [EMAIL PROTECTED]
Tobias Ahlander wrote:
Yes, I have autofill set in queues.conf. I suspect that this behaviour
is because the Polycom phones I use have 2 lines. Has
you can do the following in sip .conf file
register = username:[EMAIL PROTECTED] [EMAIL PROTECTED]
and after that write the configuration for the user:
[ user ]
username =
host =
qualify =
secret =
and so on, do this in the first of sip.conf file
Best Regards
On Mon, Sep 1, 2008 at 11:32 AM,
Michael Melia Jr. wrote:
I am looking to implement Asterisk in front of the legacy PBX as the new
gateway to the PSTN for the company in an effort to begin a slow
transition to VoIP. While I am familiar with Asterisk when it comes to
implementing it as a single box solution for PRI, SIP,
Philipp Kempgen escribió:
Dpto. Datos Television Costa Blanca schrieb:
This is my first post here and I searched a lot for a solution without luck.
Heres the problem; When I make a call forwarding from a extension to an
external number (cell phone) it never work. Only work if the
Dear,
do u have any idea to playback a remote file (with url address) ?
for example :
exten = _X.,1,playback(http://www.test.com/test.gsm;);
best
Mani
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asterisk-1.4.21.2
libpri-1.4.7
zaptel-1.4.11
I might be being a muppet here (not used PRI with Asterisk before) so
humor me.. I am using SetCallerPres on an outbound call over PRI...
Console shows:
-- Executing [EMAIL PROTECTED]:8] SetCallerPres(SIP/XXX.XXX.
209.243-08b81d68,
On 2 Sep 2008, at 10:36, Steven Howes wrote:
asterisk-1.4.21.2
libpri-1.4.7
zaptel-1.4.11
I might be being a muppet here (not used PRI with Asterisk before) so
humor me.. I am using SetCallerPres on an outbound call over PRI...
Console shows:
-- Executing [EMAIL PROTECTED]:8]
Dpto. Datos Television Costa Blanca schrieb:
Philipp Kempgen escribió:
Dpto. Datos Television Costa Blanca schrieb:
This is my first post here and I searched a lot for a solution without luck.
Heres the problem; When I make a call forwarding from a extension to an
external number (cell
Hi,
it seems zaptel 1.2.27 was tagged on 20/08 but not yet realeased in
downloads.digium.com, is there any known problem in this code that is
delaying the release ?
Leonardo
Mensagem original
Assunto: [zaptel-commits] kpfleming: tag 1.2.27 r4502 - in
/tags/1.2.27:
On Mon, 1 Sep 2008 22:01:19 -0400, Michael Melia Jr. said:
I am looking to implement Asterisk in front of the legacy PBX as the new
gateway to the PSTN for the company in an effort to begin a slow
transition to VoIP. While I am familiar with Asterisk when it comes to
implementing it as a
Hi List;
I see and hear about the Trunk version, and sometimes when I ask about
something (like media timeout for SIP trunk), then they say ur asterisk vesion
should be trunk version.
What is the difference between Trunk version and not Trunk version? And how can
I obtain the Trunk version?
bilal ghayyad wrote:
Hi List;
I see and hear about the Trunk version, and sometimes when I ask about
something (like media timeout for SIP trunk), then they say ur asterisk
vesion should be trunk version.
http://www.asterisk.org/developers/get-source
What is the difference between
Hi,
checkout
http://svnbook.red-bean.com/en/1.4/svn.tour.importing.html#svn.tour.importing.layout
this explains about versioning
Dan
On Tue, Sep 2, 2008 at 3:56 PM, bilal ghayyad [EMAIL PROTECTED] wrote:
Hi List;
I see and hear about the Trunk version, and sometimes when I ask about
Tom Moore schrieb:
What are your suggestions to people who have pbx systems that interface with
the world over pri and want to convert them to sip interfaces so that they
can use sip trunking?
I'd go for a Patton SmartNode. See www.patton.com - they have SIP
gateways up to 4 T1/E1.
Christian
On Mon, Sep 01, 2008 at 10:30:18PM +0200, Benny Amorsen wrote:
If you normally read this list via gmane and only keep a subscription
to be able to post, it's an infinite increase. Yes, I could write a
rule to junk those mails, but they aren't really consistent enough
between mailing lists.
Doug Lytle schrieb:
bilal ghayyad wrote:
What is the difference between Trunk version and not Trunk version? And how
can I obtain the Trunk version?
Trunk, is the latest 'bleeding edge' version of the software (Not yet
released).
Branch,
The current 1.4 branch ...
on the other hand,
On Tue, Sep 2, 2008 at 8:17 AM, Karl Fife
[EMAIL PROTECTED] wrote:
On Mon, 1 Sep 2008 22:01:19 -0400, Michael Melia Jr. said:
I am looking to implement Asterisk in front of the legacy PBX as the new
gateway to the PSTN for the company in an effort to begin a slow
transition to VoIP. While I
Hi
i have problem with AddQueueMember logic.
I need login Agent(Member) in asterisk.
use this option:
for example:
AddQueueMember(queuetest,SIP/ekiga,10,,Agent/13)
and now i want to call to this Agent:
exten = _1XX,1,Dial(Agent/${EXTEN:1})
call to 113 and asterisk should call to Agent =
Sorry, but I did not find in the below link anything answering the difference
between the trunk and not trunk version? When to use asterisk trunk and
asterisk normal?
Regards
Bilal
--- On Tue, 9/2/08, Dan Julius [EMAIL PROTECTED] wrote:
From: Dan Julius [EMAIL PROTECTED]
Subject: Re:
Although the original topic of this thread has changed quite a bit, I wanted
to point out that the SPF Product that you are discussing is quite similar
to our product, the FSV-4PFS. Ours is a 4 port device which can switch 4
T1/E1/J1/Ethernet or as many as 16 analog lines from a primary to a
Bilal - I think you're perhaps confusing two meanings of the word
trunk. In this case, trunk is referring to the trunk of the SVN
development repository, not SIP or IAX trunks. This can be seen as the
main development area for asterisk.
On Tue, Sep 2, 2008 at 10:22 AM, bilal ghayyad [EMAIL
Yes I mean the trunk for the development, when I have to select such version
and when I can use the normal?
Regards
Bilal
--- On Tue, 9/2/08, Erik Anderson [EMAIL PROTECTED] wrote:
From: Erik Anderson [EMAIL PROTECTED]
Subject: Re: [asterisk-users] Asterisk Trunk and normal
To: [EMAIL
Usually you'd only need to go to the trunk to get features that
haven't made it into the stable tarballs yet.
On Tue, Sep 2, 2008 at 10:37 AM, bilal ghayyad [EMAIL PROTECTED] wrote:
Yes I mean the trunk for the development, when I have to select such version
and when I can use the normal?
On Tue, Sep 2, 2008 at 11:44 AM, Pezhman Lali [EMAIL PROTECTED] wrote:
Dear,
do u have any idea to playback a remote file (with url address) ?
for example :
exten = _X.,1,playback(http://www.test.com/test.gsm;);
best
Mani
No direct way, however you can always download and play then
On Tue, Sep 2, 2008 at 6:07 PM, Krzysztof Zimnicki [EMAIL PROTECTED] wrote:
Hi
i have problem with AddQueueMember logic.
I need login Agent(Member) in asterisk.
use this option:
for example:
AddQueueMember(queuetest,SIP/ekiga,10,,Agent/13)
and now i want to call to this Agent:
exten =
On Sun, Aug 31, 2008 at 1:45 AM, C F [EMAIL PROTECTED] wrote:
No, in the beginning you asked because you don't have the experience
so folks like myself that do have the experience answered. It might
work for you, no one knows and you THINK it will work, it's a hit and
miss, stability is huge
Dear asterisk-users@lists.digium.com, Best Price Only Today.
http://byz.domemax.com?itg
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Hello,
I'm new to asterisk and i'm having a really good time configuring it.
I'd like to VoIP-to-PSTN call my SIP number (${MYSIP}) first and then
my cell phone (123456) and then finally to my voicemail.
Here's my dialplan.
exten = s,1,Answer()
exten = s,n,Dial(${MYSIP},20)
exten =
On Mon, Sep 1, 2008 at 5:57 AM, daniele visaggio
[EMAIL PROTECTED] wrote:
2008/9/1 Steven Howes [EMAIL PROTECTED]
Sip debug please.
---
Sep 1 11:53:42 VERBOSE[3599] logger.c:
-- SIP read from 10.1.1.11:5060:
SIP/2.0 405 Method not allowed
From: ;tag=as411269a4
To:
Steve Repo wrote:
Hello,
I'm new to asterisk and i'm having a really good time configuring it.
I'd like to VoIP-to-PSTN call my SIP number (${MYSIP}) first and then
my cell phone (123456) and then finally to my voicemail.
Here's my dialplan.
exten = s,1,Answer()
exten =
Hi,
On Tue, Sep 02, 2008 at 03:22:21PM -0400, John Novack wrote:
Steve Repo wrote:
Hello,
I'm new to asterisk and i'm having a really good time configuring it.
I'd like to VoIP-to-PSTN call my SIP number (${MYSIP}) first and then
my cell phone (123456) and then finally to my
Your extensions.conf looks familiar, are you using trixbox?
and are you using the web interface to configure trunk and call forwards?
if you do please post the config of your outbound route and call forward.
ron
Philipp Kempgen wrote:
Dpto. Datos Television Costa Blanca schrieb:
Philipp
Hi, all. I have a Sangoma A104D (on-board, DSP-based echo can); I'm
currently passing through some of my in-bound calls to a legacy PBX (which
I hope to eventually replace). That being said, until I do, I'd like to
kill echo cancellation for the passed-through calls -- I don't want to
mess with
Mark Spencer on TWiT's FLOSS Weekly with Leo LaPorte Randal Schwartz
This posted a few days ago. It's pretty general but Mark is in great
form.
http://twit.tv/floss38
Michael
--
Michael Graves
mgravesatmstvp.com
http://blog.mgraves.org
o713-861-4005
c713-201-1262
sip:[EMAIL PROTECTED]
skype
On Fri, 2008-08-29 at 10:26 +0530, Hiren Mistry wrote:
Hi ,
I have check zapte.conf in and after make some correction that problem
solve.
But now I am facing other problem. We are using here Postgres Database
and the data from CLI it can't insert in Postgres Database. I have
also here
I need login Agent(Member) in asterisk.
use this option:
for example:
AddQueueMember(queuetest,SIP/ekiga,10,,Agent/13)
Just out of curiosity, where do you get this AddQueueMember syntax from?
http://www.voip-info.org/wiki/view/Asterisk+cmd+AddQueueMember
Description:
sean darcy wrote:
sean darcy wrote:
sean darcy wrote:
iax.conf:
[nhi] ; receives calls
type=friend
secret=password
context=longdistance
qualify=yes
trunk=yes
callerid=test 447
extensions.conf:
[longdistance]
exten =_1NXXNXX,1,Answer()
exten
Hi,
I have been testing dundi setup, one thing i am having problem with is that
extensions are getting remove from the regcontext.
does it get removed when registration expires? how can i make sure it's added
back without power cycling the phone? which would be better, making expiration
On Tue, Sep 2, 2008 at 6:16 PM, Ken D'Ambrosio [EMAIL PROTECTED] wrote:
Hi, all. I have a Sangoma A104D (on-board, DSP-based echo can); I'm
currently passing through some of my in-bound calls to a legacy PBX (which
I hope to eventually replace). That being said, until I do, I'd like to
kill
On Sat, Aug 30, 2008 at 12:17 PM, Shariq Khan [EMAIL PROTECTED] wrote:
When i dial out any number through PRI it gives the following error every
time, while incoming calls works fine
I have sangoma E1 PRI card.
The output of a
CLI pri intese debug
at Asterisk CLI before make a test call
2008/9/1 Karl Fife [EMAIL PROTECTED]
So this card has interesting price position, the main drawback being,
IMHO,
it's eating a slot, which can be a rare resource in rackable servers.
You raise a very important point. This device uses a BRACKET, but not a
motherboard SLOT.
In other
Hello,
Receptionist is currently using an hardphone which has what I call a Last
In First Out behaviour :
whenever a new call comes in, menus that appear in phone LCD relates to the
latest incoming call not to the on-going call (if another was already
established).
So if the receptionnist was in
So it seems we've got a first successful experience with 1.6.
Are there any other ?
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