I'm sorry, I think I'm not being clear.
I want to know, for asterisk to work at all (i.e. for it to be able to connect
calls to landlines, cell phones, make outgoing calls) do I need anything beyond
a server and internet connection? Do I need a gateway of some sort? Do i need
to either have a
On 30/04/2009 6:52 p.m., don rhummy wrote:
I'm sorry, I think I'm not being clear.
I want to know, for asterisk to work at all (i.e. for it to be able to
connect calls to landlines, cell phones, make outgoing calls) do I need
anything beyond a server and internet connection? Do I need a
Hi Robert,
Please take a look at DID World Wide (www.didww.com). They have prefix 52-81
numbers.
Regards,
Gideon
From: robert.augus...@linqone.com
To: asterisk-users@lists.digium.com
Date: Tue, 28 Apr 2009 21:39:33 -0400
Subject: [asterisk-users] I am looking for a good source of Monterrey
Hello everybody.
I have a problem with an integration between an Asterisk (1.4.24.1) on
FreeBSD 7.0 and a Shoretel 7.5 server.
To make a very long story short, when someone behind asterisk call an
extension behing shoretel everything work as expected. When someone
behing the shoretel server
On 30 Apr 2009, at 04:41, Martin wrote:
No more questions. This all can be done in 2-3 hrs [PERIOD].
Then do it.
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Hello All,
Hope you all are fine and good ...
I m facing an odd problem. I am able to dial some local and Mobile Number
and some are not working.
I also try to remove and re create dial plans , but it's not working.
I live in New Delhi and Area Code is {011} but i am bale to call the number
Hello,
I am trying to enable call forwarding feature on asterisk 1.6.0.9 with
asterisk-gui. Sure there is no menu for that on gui but, when i try to
write some example scripts to extensions.conf to make it work. I totally
failed.
I dont wanna install smthing like freepbx etc on the system so, i
OK, but I do need a VOIP provider, then, right? Not just an internet provider?
And is it a special kind of VOIP you have to sign up for or can any VOIP
provider/program fulfill the needs?
Thanks!
--- On Thu, 4/30/09, Matt Riddell li...@venturevoip.com wrote:
From: Matt Riddell
Don't know that this would actually work, but Asterisk comes
out-of-the-box with 20 parking lots (701-720). If you set up a second
parking extension that started at 711 instead of 701, that would sort-of
solve your problem. An easier solution would be to send your on-hold users
to a conference
On Thu, Apr 30, 2009 at 3:07 PM, don rhummy donrhu...@yahoo.com wrote:
And is it a special kind of VOIP you have to sign up for or can any VOIP
provider/program fulfill the needs?
Depending on where you are in the world, you might want to look at
Junction Networks, VoipTalk.co.uk or Teliax.
r
Voice over Internet Protocol (VOIP) using Asterisk, Sameer Verma
work on the Programming Party to help get our own Asterisk VOIP
conference server working. :)
==
Join with the friendly productive Global FreeSW HW Culture community,
in the TWICE monthly, Voice over internet Global Conference:
Thanks.
If I already have VOIP, can I use them or is it a special kind of service I'd
need?
--- On Thu, 4/30/09, randulo spamsucks2...@gmail.com wrote:
From: randulo spamsucks2...@gmail.com
Subject: Re: [asterisk-users] What do I need to connect landline calls
without telephony
Contacttel.com , simply to use the peer option and forward all calls.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of don rhummy
Sent: April-30-09 9:49 AM
To: Asterisk Users Mailing List - Non-Commercial
In the sentence
If I already have VOIP, can I use them or is it a special kind of service I'd
need?
explain already have voip ?
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Can you setup a second parking extension? In features.conf, it lists one,
but I don't know how you would add a second one. It seems that * just kind of
makes the parkedcalls context and there isn't a way to create another one. I
could be wrong though.
[general]
parkext = 700
Just set up 6 conference
rooms and transfer the callers to the room instead of the lot. Use hints to
monitor the available rooms with a web page or asterisk managere.
I thought about that, but is there a way to pull a user out of a conference
room? Or once you parked them, would you just have
Hello,
I am trying to enable call forwarding feature on asterisk 1.6.0.9 with
asterisk-gui. Sure there is no menu for that on gui but, when i try to
write some example scripts to extensions.conf to make it work. I totally
failed.
I dont wanna install smthing like freepbx etc on the system
That's over my pay grade. Being on 1.2 makes this a little more
problematic, but there are some folks on this list who are 1.2 gurus. The
conference room idea just seemed like a good workaround since I'm reasonably
certain that the parking/features does not support a second set of lots
without a
Anyone from the asterisk list going to this?
I'm not because I don't have a commercial reason to be there to fund my
ticket but do you want to organize an Asterisk get together after the
event or the night before etc?
Regards,
Dean Collins
Cognation Inc
d...@cognation.net
+1-212-203-4357
Have you taken a look at :
http://www.freeswitch.org/asterisk_stuff/app_valetparking_1.2.c
app_valetparking for Asterisk® 1.2 is a better call parking subsystem enabling
you to have multiple 'parking lots' to place calls into, and better control on
receiving calls from
J.
-Original
Yeah, it is looking for a file parking.h which doesn't exist anywhere on my
system that I can find. Don't know what it is or where to find it.
- Original Message -
From: Jimmy Godbout s...@inbox.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
hi
I am getting this error:
-- Executing [...@smvoice-sip:1] Answer(SIP/440-0856dd70, ) in
new stack
-- Executing [...@smvoice-sip:2] rtsp(SIP/440-0856dd70,
rtsp://192.168.1.175/img/video.sav) in new stack
[Apr 30 11:22:48] WARNING[8031]: app_rtsp.c:1037 rtsp_play: rtsp play
[Apr 30
That's an interesting thought. I can figure out how to put a call on an
extension, play moh and then transfer back to the original extension if there
is no answer after say 45 seconds, but what I can't figure out is how to pickup
the call. Say I create x80 that plays moh and just waits for 45
Hello,
I am using an SPA3102, all is working with asterisk 1.4, voice mail,
outbound calling etc, and it even passes the cid name/num to my analog
phone. However, when someone is calling me, I hear the beeps but the
caller-id information is not showing up on my phone, is this an SPA3102
Hello all,
What is the recommended way to remove spaces in the name of the caller ID?
Justin.
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Can you tell me which file is looking for parking.h ? I don't see any mention
of it in the source code.
-Original Message-
From: pe...@networkoblivion.com
Sent: Thu, 30 Apr 2009 10:18:02 -0500 (CDT)
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] 2nd Parking Lot
On Thursday 30 April 2009 09:37:50 Oguzhan Kayhan wrote:
Ok I found an example script that said to be work..
but i have some errors.
Here is the script and then the error msgs.
exten = *666*,2,GotoIf($[${DB(CFBOOLEAN/${CALLERID(NUM)})} = 1]?3:102)
The DB returns nothing, so it evaluates $[ =
Newest wanpipe (3.3.16) beta drivers do not compile against dahdi-linux
2.2.0-rc2 which is what you get when you get dahdi-linux-current.tar.gz
Anyone have a workaround or patch?
Error below
Building modules, stage 2.
MODPOST
CC
Hello, I've started to do some research into the new 4G wireless
standard, and there's one part of the standard that intrigues me.
Apparently all data is packet based, including the phone calls. Every
phone will have its own IPv6 address. This seems to pave the way for
a call to go
I had a different app_valetparking.c from here:
http://www.loligo.com/asterisk/misc/apps/app_valetparking.c
There appear to be several versions floating around and it's hard to tell which
version replaces which and which one is newer.
- Original Message -
From: Jimmy Godbout
I've read a lot of conflicting information on this around the web, and
wanted to see if I could get some thoughts for any of you..
What's the proper (or best, etc) build order for install Asterisk and
it's needed libraries. Most often I see 1. Zaptel / Dahdi 2. libpri
3. Asterisk. However,
Hello,
I had same problem.
Try: ftp://ftp.sangoma.com/linux/custom/Marc/wanpipe-3.3.16.22.1.tgz.
Good luck :)
On Thu, Apr 30, 2009 at 7:10 PM, Jeremy Mann jm...@txhmg.com wrote:
Newest wanpipe (3.3.16) beta drivers do not compile against dahdi-linux
2.2.0-rc2 which is what you get when you
I found this option in asterisk 1.6.1
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Hello Jonathan,
I think the best way is:
1- Zaptel / DAHDI
2- LibPRI
3- Asterisk
On Thu, Apr 30, 2009 at 7:37 PM, Jonathan Moore supermegat...@gmail.comwrote:
I've read a lot of conflicting information on this around the web, and
wanted to see if I could get some thoughts for any of you..
Dear Aman,
You may create your own dialplan and you can try with it.
Put something like that:
exten = 611,1,Dial(ZAP/port/0114454212111|60)
then asterisk -rx 'dialplan reload', then call 611.
I have no idea about Trixbox, it's Asterisk :)
On Thu, Apr 30, 2009 at 2:50 PM, Aman Dhally
On Thu, 30 Apr 2009, Chris Kairalla wrote:
Hello, I've started to do some research into the new 4G wireless
standard, and there's one part of the standard that intrigues me.
Apparently all data is packet based, including the phone calls. Every
phone will have its own IPv6 address. This
On Thu, Apr 30, 2009 at 11:37:32AM -0500, Jonathan Moore wrote:
I've read a lot of conflicting information on this around the web, and
wanted to see if I could get some thoughts for any of you..
What's the proper (or best, etc) build order for install Asterisk and
it's needed libraries.
On Thursday 30 April 2009 10:45:48 Justin Piszcz wrote:
What is the recommended way to remove spaces in the name of the caller ID?
You could use the FILTER function, something along the lines of
Set(CALLERID(name)=${FILTER(ABCDEFGHIJKLMNOPQRSTUVWXYZ,${CALLERID(name)})})
In 1.6.0 and later, you
Hello Justin,
You can try with a softphone first.
Good luck.
On Thu, Apr 30, 2009 at 6:37 PM, Justin Piszcz jpis...@lucidpixels.comwrote:
Hello,
I am using an SPA3102, all is working with asterisk 1.4, voice mail,
outbound calling etc, and it even passes the cid name/num to my analog
Hello.
If you configure and install Asterisk first, how it detects if Zaptel or
DAHDI installed?
Thanks.
On Thu, Apr 30, 2009 at 8:15 PM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote:
On Thu, Apr 30, 2009 at 11:37:32AM -0500, Jonathan Moore wrote:
I've read a lot of conflicting information
2009/4/30 Justin Piszcz jpis...@lucidpixels.com
Hello,
I am using an SPA3102, all is working with asterisk 1.4, voice mail,
outbound calling etc, and it even passes the cid name/num to my analog
phone. However, when someone is calling me, I hear the beeps but the
caller-id information is
Had an inbound email server issue, just double checking it is working
again.
James Shigley
Monroe Telephone Answering Service
409-981-9213
Infinity 5.5,UC 4.02.3803, Blink 3.0.104
Ecreator:2.21, eResponse 1.1.7
Webportal,WebApps,
CONFIDENTIALITY NOTICE: This email, including any
Yes, its working :)
Jai Rangi
ww.didforsale.com
On Thu, Apr 30, 2009 at 12:12 PM, James A. Shigley
j...@answeringserv.comwrote:
Had an inbound email server issue, just double checking it is working
again.
James Shigley
*Monroe Telephone Answering Service*
409-981-9213**
Infinity
According to my IAX-provider, an account has been created for me on
their Asterisk-server...
But the Asterisk CLI tells me this :
asterisk*CLI iax2 reload
== Parsing '/etc/asterisk/iax.conf': Found
[Apr 30 20:51:30] NOTICE[6391]: chan_iax2.c:10124 set_config: Ignoring
bindport on reload
[Apr
In article ef4e56e70904301038u780e0d15nc94a060d72ff3...@mail.gmail.com,
Hakan C ella4e...@gmail.com wrote:
On Thu, Apr 30, 2009 at 8:15 PM, Tzafrir Cohen
tzafrir.co...@xorcom.comwrote:
On Thu, Apr 30, 2009 at 11:37:32AM -0500, Jonathan Moore wrote:
I've read a lot of conflicting
On 1/05/2009 3:31 a.m., Jerry Geis wrote:
hi
I am getting this error:
-- Executing [...@smvoice-sip:1] Answer(SIP/440-0856dd70, ) in
new stack
-- Executing [...@smvoice-sip:2] rtsp(SIP/440-0856dd70,
Best to ask in the Asterisk-Video mailing list as rtsp is not provided
in the
Hello,
I'm having an issue with caller ID in voicemail that I'd appreciate
any input on.
I have two sip peers defined as extension 100 and 101 each with
separate voicemail accounts. Each sip peer has its own DID number,
which is established via cid_number = 6021231234.
When a call is
Steve,
On Thu, Apr 30, 2009 at 5:05 AM, Steve Howes st...@geekinter.net wrote:
On 30 Apr 2009, at 04:41, Martin wrote:
No more questions. This all can be done in 2-3 hrs [PERIOD].
Then do it.
Then pay me $500
Also I see from your previous posts you like to send your little
useless
If I already have VOIP, can I use them or is it a special kind of service I'd need?
explain "already have voip" ?
I debated chiming in on this conversation. Your subject asking the same question as the text of your email. Basically you install / setup your Asterisk server. Once installed you
OK, but I do need a VOIP provider, then, right? Not just an internet provider?
And is it a special kind of VOIP you have to sign up for or can any VOIP provider/program fulfill the needs?
Most consumer VoIP providers will not meet your needs. I would strongly suggest you find a provider that
Hello, I've started to do some research into the new 4G wireless
standard, and there's one part of the standard that intrigues me.
Apparently all data is packet based, including the phone calls. Every
phone will have its own IPv6 address. This seems to pave the way for
a call to go
Is there something that you need to do so Asterisk will compile to 64bit
or will Asterisk just function as 32bit on a 64bit platform?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of sean darcy
Sent: Sunday, 26
When I compile the source code on a 64bit system it made a 64bit executable.
I did not have to do anything special. This was on a CentOS 5.3 system.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
From: Klaverstyn, David C david.klavers...@intergraph.com
Reply-To:
There is a software called Fring that use directly from my Nokia E66
phone. Fring not only connects to Skype, GMail, MSN, Yahoo, but also
allows me to register to my SIP based server and make VoIP calls over
the telco's data service. In my country I pay around $15 per month for
an unlimited data
Any ideas how to make BLF work as they should ? Just downloaded the admin
guide for SPA series
Thanks
G
On Wed, Apr 29, 2009 at 1:38 PM, Eric Chamberlain e...@rf.com wrote:
On Apr 28, 2009, at 11:36 PM, Gondar Monn wrote:
Anyone have used one of the new Cisco SPA525G with Asterisk ?
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