From: pedro noticioso [EMAIL PROTECTED]
Date: Fri, 18 May 2007 20:36:59 -0700 (PDT)
hi there!
I have a couple phones connected to a sipura ata and
if I go into *- IVR, I press options on the regular
phones and it all works fine and dandy.
then I connect an xten softphone, a new extension in
my
Brad Templeton wrote:
On Fri, May 11, 2007 at 11:06:35AM -0400, Mike wrote:
Hi,
I have a question of using 2 SIP providers. Let's say I have provider A and
provider B, and I would like my calls to go to A, and then B if A wasn`t
available
there is a macro floating around called
I have a system that has had 5 G729 licenses for over a year and I've
come to install the v31 G729 codec from the Digium ftp server but it
won't see the license. Does anyone know how to get around this problem?
It is registered and I do have newer systems running this v31 version of
the codec but
Hello Lee,
Thanks a lot thats right but in i hearing tone when i click on buton but it
not take asterisk as a DTMF generate code so voice mail not identified.
thats problem . if u knw then reply me.
Regards,
gaur
On 5/18/07, Lee Jenkins [EMAIL PROTECTED] wrote:
gaurang sheladiya wrote:
Hi,
I am faced with this dilema of asterisk not sending an ACK after it receives
200 OK from OpenSER (which is a response to a reinvite request sent by
asterisk. Here is my setup
Carrier-OpenSER-Asterisk1-Asterisk2
A user is connected with Asterisk1 (through the carrier and OpenSER). On
With all of the recent talk on the list about DUNDi, I have a question. From
the outset it appears that SER is often used for high availability solutions
and as a tool for almost clustering Asterisk boxes behind it. It appears to
me that DUNDi is providing a lot of this as well. Now I know DUNDi
dear
is any snmp access , for asterisk 1.2.* ?
Boardwalk
for $500? In 2007? Ha! Play Monopoly Here and Now (it's updated for today's
economy) at Yahoo! Games.
You can just do it in the dialplan without changing an code.
Thanks,
Steve Totaro
http://www.asteriskhelpdesk.com
KB3OPB
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Brad Templeton
Sent: Saturday, May 19, 2007 12:22 AM
To:
You heard Terrasip
Try on www.terrasip.com
Special rate list or discounts.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anthony
Francis
Sent: Wednesday, May 16, 2007 6:59 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
What is the difference between the 1.2.x and the 1.4.x branches? I am starting
to deploy asterisk from scratch. I expect that I should use the 1.4.x branch,
but don't know the rationale for the two.
+
This e-mail was checked by the TecInfo Content Scanning Service for potentially
still nothing... it gives DUNDi lookup returned no results. DUNDi
lookup completed in 0 ms
greetz
2007/5/17, JR Richardson [EMAIL PROTECTED]:
[mappings]
priv = dundi-priv-canonical,0,SIP,${IPADDR}/${NUMBER},nopartial
priv = dundi-priv-customers,100,SIP,${IPADDR}/${NUMBER},nopartial
priv =
Curt Shaffer wrote:
With all of the recent talk on the list about DUNDi, I have a question. From
the outset it appears that SER is often used for high availability solutions
and as a tool for almost clustering Asterisk boxes behind it. It appears to
me that DUNDi is providing a lot of this as
I am new at this. I have read Asterisk: The Future of Telephony and have
installed AsteriskNOW (beta 4, due to the dual processor problem in beta 5).
The GUI interface does not seem to provide the capability that I need, although
I have modified the *.conf files to successfully create what I
Tim Verscheure wrote:
still nothing... it gives DUNDi lookup returned no results. DUNDi
lookup completed in 0 ms
have you set dundi debug? Is there any communication happening? Test
with 'dundi lookup number bypass'.
greetz
2007/5/17, JR Richardson [EMAIL PROTECTED]:
[mappings]
priv
Malcom Kemp wrote:
I am new at this. I have read Asterisk: The Future of Telephony
and have installed AsteriskNOW (beta 4, due to the dual processor
problem in beta 5). The GUI interface does not seem to provide the
capability that I need, although I have modified the *.conf files to
Remco Post wrote:
Tim Verscheure wrote:
still nothing... it gives DUNDi lookup returned no results. DUNDi
lookup completed in 0 ms
have you set dundi debug? Is there any communication happening? Test
with 'dundi lookup number bypass'.
'dundi lookup number@priv bypass' of course
greetz
2007/5/19, Remco Post [EMAIL PROTECTED]:
Tim Verscheure wrote:
still nothing... it gives DUNDi lookup returned no results. DUNDi
lookup completed in 0 ms
have you set dundi debug?
yes
Is there any communication happening?
debianPBX*CLI dundi lookup 6000 bypass
DUNDi lookup returned no
OK this worked... THANK YOU SO MUCH!
So now I'm able to call?
2007/5/19, Remco Post [EMAIL PROTECTED]:
Remco Post wrote:
Tim Verscheure wrote:
still nothing... it gives DUNDi lookup returned no results. DUNDi
lookup completed in 0 ms
have you set dundi debug? Is there any communication
Hi!
I would like to connect a door phone to my asterisk server.
I decided to do that with an analog (a/b) doorphone and a sipura box.
Can anyone give me a recommendation for a door phone with good voice
quality?
I am from Austria/Europe and I looked at products from Rocom and from
Auerswald.
I tried to call to with X-Lite to extension 6000 but it still doesn't
go through:
Call Failed: Not found
greetz
2007/5/19, Remco Post [EMAIL PROTECTED]:
Remco Post wrote:
Tim Verscheure wrote:
still nothing... it gives DUNDi lookup returned no results. DUNDi
lookup completed in 0 ms
Hi,
We are currently trying to setup Asterisk with iBasis. One question/problem we
have is that Ibasis has told us to send the INVITEs to one IP address and all
media to a different IP address. How can we do that in Asterisk?
Thanks
___
--Bandwidth
Tim Verscheure wrote:
I tried to call to with X-Lite to extension 6000 but it still doesn't
go through:
Call Failed: Not found
so you'll have to set de dundi context in your switch statement as well,
else the switch will look in an e164 context, that doesn't exist.
greetz
2007/5/19,
Tim,
When you are checking for 5000 are you typing dundi lookup [EMAIL PROTECTED]
On 5/18/07, Tim Verscheure [EMAIL PROTECTED] wrote:
Thank you for the quick response. Do I need to create a route to the
other machine? like a trunk?
greetz, Tim
2007/5/17, JR Richardson [EMAIL PROTECTED]:
2007/5/19, Bruce Reeves [EMAIL PROTECTED]:
Tim,
When you are checking for 5000 are you typing dundi lookup [EMAIL PROTECTED]
Remco just told me this and indeed it worked. The lookup now returned
a result! Now I'm trying to make a call
thanks
On 5/18/07, Tim Verscheure [EMAIL
like this???
[dundi-priv-switch]
; Just a wrapper for the switch
switch = DUNDi/priv
2007/5/19, Remco Post [EMAIL PROTECTED]:
Tim Verscheure wrote:
I tried to call to with X-Lite to extension 6000 but it still doesn't
go through:
Call Failed: Not found
so you'll have to set de dundi
I have no affiliation with them but if their quotes are accurate then
they provide quite a few options as far as TDM connectivity and realtime
pricing.
If you do not want a phone call from a sales person, give them a BTN
that goes to an IVR or something. They call no matter which box you
click
Tim Verscheure wrote:
like this???
[dundi-priv-switch]
; Just a wrapper for the switch
switch = DUNDi/priv
yes that should do. Does your asterisk console show anything useful? And
if you do wind up in the switch, what does you dundi debug show?
--
Remco Post
I didn't write all this
Malcom Kemp wrote:
Has anyone put Asterisk on the 10.2 distro? Any pointers?
Yes, we're running 1.4.4 on openSUSE 10.2. We're have a couple of ISDN
lines fed into each a TA card with a Cologne HFC chip.
What you need to do is configure the before you jump to trying to build
asterisk. This
Asterisk supports it and the good news is that you don't have to do anything
for it to work.
On 5/19/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Hi,
We are currently trying to setup Asterisk with iBasis. One
question/problem we have is that Ibasis has told us to send the INVITEs to
one IP
Hello,
If properly configured, does Asterisk 1.2 'push' SIP NOTIFY messages
to registered SIP phones to indicate new voicemail?
I've noticed that Asterisk only sends NOTIFY messages when my phones
periodically (re-) register with Asterisk---rather than immediately
after a new voicemail arrives.
Thank you for the quick response. Do I need to create a route to the
other machine? like a trunk?
On the SIP side of things, yes, you can create a SIP trunk for each
server-to-server relationship, or you can just send the sip call to the
default context and use a goto statement to get the call
Thanks for the prompt response, but would you care to explain this a bit
further?
It could be due to my ignorance, and if so, I apologize. But, how can we send
the INVITE to one IP and then the media to a different one? Do we just simply
send the call to the INVITE IP using the Dial command
If I read all this is realize what a noob I am in this matter.
Could I make a call by saying something like this:
exten = 16000,1,Dial(SIP/[EMAIL PROTECTED])
Or something like that?
2007/5/19, Remco Post [EMAIL PROTECTED]:
Tim Verscheure wrote:
like this???
[dundi-priv-switch]
; Just a
Hi List
Whats the best way to run * on Solaris 10 with x86 architecture. I am
following solarisvoip.com using svn, but came across issues like
1. app_lookupcnam compilation issue - Wrong format of ELF.
Is this the correct way.
___
--Bandwidth
With all of the recent talk on the list about DUNDi, I have a question.
From
the outset it appears that SER is often used for high availability
solutions
and as a tool for almost clustering Asterisk boxes behind it. It appears
to
me that DUNDi is providing a lot of this as well. Now I know
VoIP User wrote:
Hi Everyone,
can you please recommend me a good VoIP provider as I am not satisfied
by my current provider. Does not matter what protocol it uses. I'm
looking for good rates, stable quality and not so big prepayment required.
Thanks to all.
Hi all.
When i do a service zaptel stop on my machine,sometimes it crash and i
must unplug and plug the power cord to restart the machine. Also sometimes
load zttranscode and wct4xxp, and oter times wct4xxp only... it's running
centos 4.4, libpri 1.2.4, zaptel 1.2.17.1 and asterisk 1.2.18. with a
We do that.
www.direct-internet.co.in. BTW whats your location and where is your
major calling.
Lee Jenkins wrote:
VoIP User wrote:
Hi Everyone,
can you please recommend me a good VoIP provider as I am not
satisfied by my current provider. Does not matter what protocol it
uses. I'm
Tim Verscheure wrote:
If I read all this is realize what a noob I am in this matter.
Could I make a call by saying something like this:
exten = 16000,1,Dial(SIP/[EMAIL PROTECTED])
you could, look into the DUNDILOOKUP function...
Or something like that?
2007/5/19, Remco Post [EMAIL
Hi,
have you installed asterisk-addons, too?
The cdr-MySQL driver is in the addons package.
CLI show modules
...
cdr_addon_mysql.so MySQL CDR Backend
...
When this line appears, the driver is allready there.
Your cdr_mysql.conf must contain something like that:
[global]
iBasis, like many providers uses a softswitch in which separate elements
handle the signaling (SIP/H.323) and media gateways handle the media (RTP).
when you send a call with the Dial command you state iBasis signaling
address and the Asterisk sets it's own media IP/Port in the SDP. when iBasis
It should work the way you outlined; if your running through a
firewall just make sure you allow traffic to/from both of the
provider's IP addresses.
On 5/19/07, Yossi Ben Hagai [EMAIL PROTECTED] wrote:
iBasis, like many providers uses a softswitch in which separate elements
handle the
Thank you all for the responses.
Just one last question :)
We get the call from another VoIP provider. Once we validate the call (e.g.
calling card), we send the call to iBasis and we don't want to stay in the
media path. So, we have canreinvite=yes. This works for us for other providers.
So,
I think the best way is to conact Digium Hardware support. it seems there may
be an IRQ problem.
--
Deepak
Francois Deppierraz [EMAIL PROTECTED] wrote:
Hi,
I'm trying to get a TE212 working on a Dell PowerEdge 1850 running
Debian etch using the latest release of libpri (1.4.0), zaptel
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Danish Samad wrote:
Hi,
I am faced with this dilema of asterisk not sending an ACK after it
receives
200 OK from OpenSER (which is a response to a reinvite request sent by
asterisk. Here is my setup
Firstly don't cross post.
Olle posted a fix
On Wed, 9 May 2007, Ritesh Agrawal said something to this effect:
Is there a way to find out the mobile/landline carrier name based on the
phone number?
Ordinary people can only find this out if the NPA-NXX (area code +
exchange, i.e. the first six digits) block to which the number belongs
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José Hugo Pérez Casanova wrote:
Hi Folks.
I have installed two sip phones and two PCs in a network. The later with
iaxComm. Calls are made between the sip phones and between a sip phone
and a
PC.
When calling from one PC to the other the
On 5/9/07, Ritesh Agrawal [EMAIL PROTECTED] wrote:
Hi Folks,
Is there a way to find out the mobile/landline carrier name based on the
phone number?
For example, who is the mobile carrier for (415)2345678
I had heard about some query but just don't remember how/what?
Thanks in advance.
Ritesh
Malcom:
Great to know there are more loyal SuSE users like myself!
After you install your kernel source have you tried:
# cd /usr/src/linux
# make cloneconfig
# make prepare-all
The problem is SuSE does not provide the kernel headers, you need to
create them yourself.
Of course this assumes
I would suggest you at least look into DIALSTATUS.
Guys:
The dialplan is not a toy. You need to consider with care the results
of your actions. What Mike posted is an example of a bad dialplan in
the works.
On 5/11/07, Mike [EMAIL PROTECTED] wrote:
Hi,
I have a question of using 2 SIP
Is there a way to find out the mobile/landline carrier name based on the
phone number?
For example, who is the mobile carrier for (415)2345678
I don't know what the story is regarding number porting over the other side of
the pond, but here in the UK where number portability's been around
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Tony Plack wrote:
I believe this is a Asterisk console bug, but thought I would run it through
here first. I can get Asterisk into a tight loop 100% of the time. Here is
what I know...
Hmm, can't add much except that I use SecureCRT from
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Hash: SHA1
Carlos Chavez wrote:
On Fri, 2007-05-18 at 16:42 -0600, Anthony Francis wrote:
Use the cdr's, who wont know who but at least which phone did it.
I tried following the CDR but if I dial extension 4000 and extension
4002 picks up the call
Hi,
I was just wondering how would the application be where the Asterisk calls a
number and that number joins the conference as soon as the call connects.
There would be only one conference already defined in meetme.conf and there
is one person already joined the conference. Currently MeetMe
Dear All,
I am using asterisk+ss7+TE407P card at my box. here i have 4 E1 and only
one signal channel. problem is that my 1st E1 work fine. but from 2 to 4th
E1 i have got only ring tone but no voice from both side. what is the
problem. pls suggest me. here is my conf file
zaptel.conf
Any help is appreciated.
Kapil Dhawan wrote:
Hi List
Whats the best way to run * on Solaris 10 with x86 architecture. I am
following solarisvoip.com using svn, but came across issues like
1. app_lookupcnam compilation issue - Wrong format of ELF.
Is this the correct way.
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