How can I force my users to be obliged to give a 'fromuser' and
'fromdomain' -parameter in their SIP-configuration ??
Is this set in the [general] -section of sip. conf ??
Jonas.
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Dear All,
I have installed Asterisk 1.6.1.9 to use Bridge Application in AMI.
After inatallation I have tried to connect the AMI via telnet. But it
didn't connected. I used netstat to know the listening socket. But it was
not available. How to start the AMI server socket.
Please any one
Hi,
I have a digium card (igium, Inc. Wildcard TE405P quad-span T1/E1/J1 card
5.0V (rev 02)) 4 ports
I want to make a loop test between digium card E1 to test the
configuration of dahdi
What I want to do scenario is
I connect port 1 and port4 in the digium card with E1 cable
SIPcall--E1
Find my dahdi config files below
dahdi-channels.conf
; Span 1: TE4/0/1 T4XXP (PCI) Card 0 Span 1 (MASTER) HDB3/CCS/CRC4
ClockSource
group=0,11
context=default
switchtype = euroisdn
signalling = pri_cpe
channel = 1-15,17-31
context = default
group = 63
; Span 2: TE4/0/2 T4XXP
Hi,
On Mon, Nov 09, 2009 at 12:52:15PM +0200, Khaled W Chehab wrote:
Hi,
I have a digium card (igium, Inc. Wildcard TE405P quad-span T1/E1/J1 card
5.0V (rev 02)) 4 ports
I want to make a loop test between digium card E1 to test the
configuration of dahdi
This is fairly simple. But I
velusamy velu wrote:
Dear All,
I have installed Asterisk 1.6.1.9 to use Bridge Application in
AMI. After inatallation I have tried to connect the AMI via telnet. But
it didn't connected. I used netstat to know the listening socket. But
it was not available. How to start the AMI server
Sendtext() works for SIP endpoints
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Thomas Perron
Sent: Saturday, November 07, 2009 9:39 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Text messaging
IVR
That may not work for all sip phones. Some (like xlite/eyebeam) crash when
receiving a text, others drop the subsequent call (Aastra 5x). These
observations are based on a project we did in late 2008; so be sure to do a
proof of concept before you get too deep into the project.
_
From:
What does Sendtext() actually do? Does it send a SIP request of
method MESSAGE? What does it do on a hardware channel - say, analog
or TDM?
Michelle Dupuis wrote:
That may not work for all sip phones. Some (like xlite/eyebeam) crash
when receiving a text, others drop the subsequent call
It does nothing on hardware channels.
SendText is just works on SIP channels.
Purpose of SendText is showing text messages on user phone screen.
show application SendText
-= Info about application 'SendText' =-
[Synopsis]
Send a Text Message
[Description]
SendText(text[|options]): Sends
I assumed the ATA/gateway would throw away or reject the message since I
don't think there's an analog equivalent...but I'll wait for the analog
experts to jump in.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf
Hi,
I have a digium card (digium, Inc. Wildcard TE405P quad-span T1/E1/J1 card
5.0V (rev 02)) 4 ports
I want to make a loop test between spans on digium card in order to test
the spans.
I connect port 1 and port4 with cross E1 cable
I am trying to do this scenario
SIPcall-- Digium span
Thanks for suggestions, everyone- I should have thought about jitter and
latency as I began to use up more more bandwidth. I was concerned that it
was a problem with my configuration of Asterisk, but it looks like is really
is a bandwidth issue. By the way, Joe- I've been in another situation
Michelle Dupuis wrote:
I assumed the ATA/gateway would throw away or reject the message since I
don't think there's an analog equivalent...but I'll wait for the analog
experts to jump in.
It appears that Sendtext() simply invokes the callback stub
ast_channel_tech.send_text, and this is
I am using the AMI to dispatch (2) calls to Local/my_prior...@my_context
where:
[my_context]
exten = my_priority,1,Answer()
exten = my_priority,n,Dial(${LOCAL_DIAL})
and LOCAL_DIAL has the actual phone number to dial.
The first call goes through just fine and I see DAHDI/1/ being
called.
Hakan,
I did not ask about the purpose of Sendtext() - I know the purpose,
and on the level on which you have explained it, it is self-evident.
I asked about how it was implemented underneath. Even in the context
of SIP channels solely, there are numerous ways to send what one might
term a
Jerry Geis wrote:
I am using the AMI to dispatch (2) calls to Local/my_prior...@my_context
where:
[my_context]
exten = my_priority,1,Answer()
exten = my_priority,n,Dial(${LOCAL_DIAL})
and LOCAL_DIAL has the actual phone number to dial.
The first call goes through just fine and I see
As I said, please keep discussion on list.
aster...@opensourcesolution.in wrote:
hi all,
first of all i appologise for sending u pvt email. i have installed
asterisk on Centos 5.3, plz open the attachment in which i had drawn a
tolpology. i had installed one asterisk machine and two
*Darrick Hartman:*
NO! If you're using a specific 'branch' of asterisk, the latest release
in that branch is the recommended version. There are almost certainly
bugs/issues with earlier versions. 1.6.1.9 is the recommended version
of Asterisk 1.6.1.x.
*Danny Nicholas:*
RC's are
Hi,
He did that to me too (and previously). He's a complete fucking pain.
I find it laughable that someone working for 'opensourcesolution' cant
install a damned softphone. Clearly he is in the wrong business.
Steve
On 9 Nov 2009, at 16:32, Alex Balashov wrote:
As I said, please keep
I would tend to concur.
This is not an uncommon phenomenon on these lists and especially from
that part of the world, however. People like this are not easily
discouraged by criticism nor encumbered by any interest or sensivity
in the prevalent ethics and culture of forums into which they
Hello everybody,
This is my first post to this mailing list, so welcome everybody and
thanks for the great community around asterisk.
I few weeks back I got control over an asterisk server and was asked
to create a number forwarding by the means of the configuration files.
With the help of the
hi all,
i have installed asterisk on Centos 5.3, plz i had installed
one asterisk machine and two windows machine. now i want to install
softphone in both windows machine. and both softphone should communicate
with each other. any support and guidance will be highly appreciated.
thx
Hi all,
In the INVITE from my SIP provider to Asterisk i can see that the
Allow Header includes an INFO Method, but Asterisk replies a 200 OK
with an Allow Header without INFO Method. But in the RFC3261 (20.5)
you can read:
All methods, including ACK and CANCEL, understood by the UA MUST be
That's what yahoo.answers.com is for!
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alex Balashov
Sent: Monday, November 09, 2009 11:23 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc:
You just don't get it, do you?
Your indolent methods of getting what you want are not at your
disposal here.
This is not a homework help forum.
--
Sent from mobile device
On Nov 9, 2009, at 12:11 PM, aster...@opensourcesolution.in wrote:
hi all,
i have installed asterisk on Centos 5.3,
Yes, it's correct. Asterisk needs to advertise its support of that
method in order for the other UA to be willing to send messages with
that request method to it.
Coco Richard wrote:
Hi all,
In the INVITE from my SIP provider to Asterisk i can see that the
Allow Header includes an INFO
I currently have an Asterisk running on an Alix 6B2 from PC Engines, but I am
having trouble using ztdummy as a timing device. The USB driver is OHCI, and
I believe ztdummy requires UHCI.
So, I am wondering if there is a way to use a Kernel tick and ztdummy on
FreeBSD, like it is possible on
Hello all,
I have successfully paired my mobile with asterisk, and chan_mobile
already run very well, but sometimes when i restart asterisk chan_mobile
fails to initialize with the error:
chan_mobile.c: Incorrect voice setting for adapter toshiba, it must be
0x0060 - see 'man hciconfig' for
Jerry Geis wrote:
/ I am using the AMI to dispatch (2) calls to Local/my_priority at
my_context http://lists.digium.com/mailman/listinfo/asterisk-users
// where:
// [my_context]
// exten = my_priority,1,Answer()
// exten = my_priority,n,Dial(${LOCAL_DIAL})
//
// and LOCAL_DIAL has the
I think the problem is that the way this works - if I'm not mistaken -
is that the attribute after the first delimeter in the channel string
is a trunk group and not a channel.
In other words, DAHDI/1 refers to circuit 1, not B-channel 1 of
circuit 1. B-channel 1 would be DAHDI/1/1.
Jerry
My Dial() command is Dial($LOCAL_DIAL)
Perhaps you should be using:
Dial(${LOCAL_DIAL})
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asterisk-users mailing list
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/ My Dial() command is Dial($LOCAL_DIAL)
/
Perhaps you should be using:
Dial(${LOCAL_DIAL})
Steve,
Thanks I tried that also and same result.
Jerry
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Is there a way to tell if an extension is in use? We run a call center and it
would be helpful for the people taking calls to see if we are on the phone or
DND. Is that setup in Asterisk or on the phone? the phone as busy lamp field
but i will just turn on after a while even if the extension
This is what I see:
-- Executing [my_prior...@my_context:1]
Answer(Local/my_prior...@my_context-90d5,2, ) in new stack
-- Executing [my_prior...@my_context:2]
Dial(Local/my_prior...@my_context-90d5,2, DAHDI/3/4000) in new stack
[Nov 9 16:25:17] WARNING[8979]: app_dial.c:1275
You can use hints to tell If a line is inuse. There are built-in functions
that do this also, but they don't always produce the desired result
depending on what release you are on.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On
LOCAL_DIAL is populated
- exten = s,1,Verbose(call ${LOCAL_DIAL})
- exten = s,2,Dial(${LOCAL_DIAL})
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jerry Geis
Sent: Monday, November 09, 2009 3:17 PM
To:
So 4001 is a local FXS DAHDI channel?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jerry Geis
Sent: Monday, November 09, 2009 3:33 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
Hi Alex,
i'm not sure to understand. Asterisk does support SIP INFO, so why
doesn't Asterisk add the INFO Method in the 200OK Response?
richard
On Mon, Nov 9, 2009 at 6:38 PM, Alex Balashov abalas...@evaristesys.com wrote:
Yes, it's correct. Asterisk needs to advertise its support of that
Dear all,
I'm in basic setup of my network:
I try to do a call from a softphone to an other one but I got the error 603
Declined.
Below the
sip.conf:
*[gianca]
type=friend
username=gianca
secret=pwd_gianca
host=dynamic
context=tutorial*
*[giusy]
type=friend
username=giusy
secret=pwd_giusy
Try:
[tutorial]
exten = 1234,1,Dial(SIP/gianca,10,t)
exten = 12345,1,Dial(SIP/giusy,10,t)
You want a / between SIP and the name of the phone, not an ,.
The 10 refers to the number of seconds you want the phone to ring. The t
allows the channel to be transferred after pickup - not strictly
Does anyone have any success with sending a text message from
extensions.conf
to an PSTN endpoint such as a cell phone?
If so, kindly send configuration for this part. I am working on an IVR and
want
callers to get a text message at a particular part of the call, after
dialing a defined
Have you taken a look at the following?
http://www.astassistant.com/
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ott Rose
Sent: Monday, November 09, 2009 4:25 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users]
Hi All;
I just need to know the openion about Grandstream phone, actually I tried Budge
Tone 201 and I chocked that there is a noise in the handset
(zzz) always, but in the speaker the sound is good and
no noise.
Anyone has idea about Grandstream, and if they have a
He wrote me too. I would have helped him, but the name on the email address
threw me off.
CS
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Monday, November 09, 2009 9:56 PM
To:
On 10/11/09 1:12 PM, bilal ghayyad wrote:
Hi All;
I just need to know the openion about Grandstream phone, actually I tried
Budge Tone 201 and I chocked that there is a noise in the handset
(zzz) always, but in the speaker the sound is good
and no noise.
Anyone
Hi,
I understand that speech recognition technology is not very reliable, but
skype has has launched a voicemail to text service, and googling showed that
some other companies are also offering similar services. I haven't used any
such service yet, but was curious is there any open source
On 10/11/09 1:02 PM, Conklin, Tom wrote:
Have you taken a look at the following?
http://www.astassistant.com/
Also:
http://www.asternic.org
and the newer version:
http://www.fop2.com
--
Cheers,
Matt Riddell
Director
___
On 10/11/09 12:58 PM, Thomas Perron wrote:
Does anyone have any success with sending a text message from
extensions.conf
to an PSTN endpoint such as a cell phone?
If so, kindly send configuration for this part. I am working on an IVR
and want
callers to get a text message at a particular
On 10/11/09 4:08 AM, C. Savinovich wrote:
He wrote me too. I would have helped him, but the name on the email address
threw me off.
Poor guy - language/cultural barrier maybe?
Here's some tips:
1. Read Asterisk The Future of Telephony (buy a copy or download from
http://asteriskdocs.org)
There are a couple of ways you could see that,
One would be by having a service .NET connected to the manager interface
and watching for activity on the phone, this way you could tell if the
phone is busy or not.
[If phone has more than one line then set call-limit=1]
Is this for routing
On Sat, Nov 7, 2009 at 11:36 AM, Warren Selby wcse...@selbytech.com wrote:
I think your featureLabel definition is wrong.
On the login issue, ssh to the ip of the phone and login first with
the user/pass you defined in the file (admin/123), then at the second
login prompt use log/log. That
Will text messages work to non-SIP enpoints using your logic/code?
thank you
On Mon, Nov 9, 2009 at 8:59 PM, Matt Riddell li...@venturevoip.com wrote:
On 10/11/09 12:58 PM, Thomas Perron wrote:
Does anyone have any success with sending a text message from
extensions.conf
to an PSTN
On Monday 09 November 2009 15:38:54 Coco Richard wrote:
i'm not sure to understand. Asterisk does support SIP INFO, so why
doesn't Asterisk add the INFO Method in the 200OK Response?
You must be using Asterisk 1.2. This is the only version that I could find
that does not put the INFO tag into
Maybe, you should take a look at 1.6.1.10-rc2 published yesterday.
It includes an audiohook-memory patch which might correct the root cause of
these crashes.
As 1.6.1.9 is a security-only release, I don't think it should improve
anything (beside security fix, of course).
Regards
On 10/11/09 4:19 PM, Thomas Perron wrote:
Will text messages work to non-SIP enpoints using your logic/code?
thank you
If you mean SMS, yeah.
Basically use SendText for devices which can display them (i.e. SIP/IAX
phones) and Clickatel or the like for disconnected devices (i.e. SMS to
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