Hi All,
I know this is not specifically Asterisk related but I don't knew where
else to ask for help. Does anyone know how to or if it is even possible
to allocate 512kbit/s to an ISDN device from a 30B+D ISDN line.
The building the office is in has a E1 30 channel service (30B+D) but we
Hi,
you can generate html file from doc/tex/queues-with-callback-members.tex for
an example.
2010/4/7 Joe Freeman j...@ngn-networks.com
Since AgentCallbackLogin() was apparently removed from 1.6, does anyone
have anything to replace that functionality?
Thanks-
Joe
--
Hi All,
I am writing to you for Packt Publishing, the publishers of computer related
books.
We are planning to extend our catalogue of books on Open Source System and
Network Administration are currently inviting asterisk experts to write for
us. So, if you love Asterisk and fancy writing
On Wed, 2010-04-07 at 21:37 -0700, Steve Edwards wrote:
On Thu, 8 Apr 2010, Pham Quy wrote:
I want to have a separate file to log what i need for my dialplan
without all output from Asterisk. By this way, i can easily to trace
problems caused by my dialplan.
You can control how much
Hi,
How to set a Gigaset S450IP up to support R-key transfers ?
Mine is enabled with firmware 02223.
In Settings/Telephony/Advanced Settings, I set:
DTMF Send Settings: SIP info
Call transfer:
Use the R-Key to initiate call transfer: yes
Transfer call by on-hook: yes
Derive target address: from
I want to use Asterisk as a general message delivery system here.
That is, I want to be able to have a (shell, perl, etc.) script on my
Asterisk server dial an extension, wait for it to be answered and then
play a sound file and then hang up, or even wait for a response or
reactions to some IVR.
On Thu, Apr 08, 2010 at 07:00:11AM -0400, Brian J. Murrell wrote:
I want to use Asterisk as a general message delivery system here.
That is, I want to be able to have a (shell, perl, etc.) script on my
Asterisk server dial an extension, wait for it to be answered and then
play a sound file
AGI and AMI is what you need for this.
AMI is for originating the call between extensions
AGI for playing file of your choice.
Both these APIs are well documented
http://www.voip-info.org/wiki/view/Asterisk+AGI
http://www.voip-info.org/wiki/view/Asterisk+manager+API
--
Thanks Regards,
On Mon, Feb 8, 2010 at 2:20 AM, Olle E. Johansson oej at edvina.net
wrote:
7 feb 2010 kl. 15.09 skrev Per Jessen:
Thomas Winter wrote:
Hi,
my Asterisk on debian lenny died after 80 days.
server kernel: [7572666.186852] asterisk[3673]:
segfault at 10 ip 7f3b8e90b4aa sp 40bf5f00 error
Have a look at the call files examples of voipinfo
http://www.voip-info.org/wiki/view/Asterisk+auto-dial+out
Its not too hard to do what you want
Cheers Duncan
On 8/04/2010, at 11:00 PM, Brian J. Murrell wrote:
I want to use Asterisk as a general message delivery system here.
That is, I
Thanks guys for all the input. I have just noticed that the solution doesn't
work for me because the 20 lines are in a hunt. And the line in problem is
actually the 4th line and not the 1st. So, for incoming calls, if I have
more than 3 calls the 4th one will keep ringing for ever and it won't go
Howdy,
Can anyone point me to links or discussions about realtime jitter
measurement? I read a long thread from 2007 (Douglas Garstang) that
didn't end with any conclusions. I want to do the same thing he was
trying to do - allow realtime jitter measurements to help control call
routing with
In your zapata.conf, under group 0 do: channel = 1-3,5-20. It works for PRI
but haven't tried it for FXO.
Have you tried disabling this line in zaptel.conf? Maybe that'll help.
Zeeshan A Zakaria
--
Sent from my Android phone with K-9 Mail.
On 2010-04-08 8:29 AM, bruce bruce bruceb...@gmail.com
bruce bruce wrote:
Is there anyway I can put a busy voltage on this line without ramping
up a big bill? If the line status shows busy then both of incoming and
outgoing calls will use the next line available.
Isn't it as simple as unplugging that phone line from the card?
Doug
--
Ben
I am not sure if unplugging line from card would work as it's still in a
hunt and calls will keep coming through that number and won't fall over to
next line unless there is a BUSY on the line. There is no timeout; it's a
hunt on BUSY. Plus, I don't have site access for two days :-)
For calls out
bruce bruce wrote:
I can't check zaptel disable of the line now as it nears 9:00 A.M.
operation time. I will try that later in the day. I am amazed there is
not much control to the lines in situations like this.
Actually,
I think you could plug it into a normal phone and leave it off
On Thu, 8 Apr 2010, bruce bruce wrote:
I am not sure if unplugging line from card would work as it's still in a
hunt and calls will keep coming through that number and won't fall over to
next line unless there is a BUSY on the line. There is no timeout; it's a
hunt on BUSY. Plus, I don't
Jeff LaCoursiere wrote:
On Thu, 8 Apr 2010, bruce bruce wrote:
Nope - unplugging a line that is in a hunt will result in Ring-No-Answer.
Ditto for previous advice to destroy the zap channel or to leave it out of
Our telecom guy said, that when you call the line in for repair, that
Thanks for the input.
Yep, a busy feature on zaptel is an absolute necessary. See, this is a sort
of problem that comes back to everyone and goes away quickly, hence the
feature wasn't developed probably. But it will make a great addition and
will help people in situations like this.
On Thu,
Hi,
I just purchased an additional license from Digium but the problem is
still there.
The output g729 show licenses command when not in a call
#g729 show licenses
0/0 encoders/decoders of 2 licensed channels are currently in use
*The output *g729 show licenses command* when there is a
Hi
I have set MeetMe options like *sdMS(10)L(1000)* in dialplan.
But when i print this value in c file using ast_log.. I am getting
only *sdMS(10
*this options.
Is there any special way to set option in dialplan with *sdMS(10)L(1000) *in
dialplan
--
Regards,
Chandrakant Solanki
--
We have been experimenting with how many licenses are needed when making calls,
recording calls and using chanspy to listen in on calls when G729 is involved.
I can tell you that way more licenses are needed then I had understood
previously. We are making calls via AMI originate and both legs
All,
I am looking at a little support on this, as I haven't found it on
google yet. I have had this work on Callweaver, but am moving to
Asterisk for a variety of reasons. My dial plans, and everything else
transferred perfectly, though I am not sure they are 'correct' for
Asterisk
hi:
maybe you can use part of channels in 30B, for examle, 10 channels, and set the
rest to unused.
_
Hotmail: Powerful Free email with security by Microsoft.
hi:
i think it should be ok. if down, you can not make calls.
Best wishes!
Asterisk Support group for sangoma, digium...
websites: www.cnasterisk.com, www.voip88.com
Date: Wed, 7 Apr 2010 11:44:16 -0400
From: stot...@first-notification.com
To: asterisk-users@lists.digium.com
Subject: Re:
any clue Guys???!!!
2010/4/5 khalid touati khalidtou...@gmail.com
Hi Juan,
my system is an asterisk 1.2 on gentoo, it is configured to receive faxes
through rxfax and then to use fax2email to convert the tiff to pdf and send
it to front desk:
exten =
Jim-
We have been experimenting with how many licenses are needed
when making calls, recording calls and using chanspy to
listen in on calls when G729 is involved. I can tell you that
way more licenses are needed then I had understood
previously. We are making calls via AMI originate and
On Wed, Apr 7, 2010 at 10:12 PM, Pham Quy qu...@vega.com.vn wrote:
Hi all,
I want to have a separate file to log what i need for my dialplan
without all output from Asterisk. By this way, i can easily to trace
problems caused by my dialplan.
How can i do that?
That's honestly a pretty
What is the consensus on using the 1.4 jitterbuffer? Do most people
enable it?
We have a PSTN server that has our RBS T1 trunks in a central location,
then have clients that connect via SIP to us for access to those trunks.
Most of them are just fine, but lately we have a handful that are
- Jeff LaCoursiere j...@jeff.net wrote:
What is the consensus on using the 1.4 jitterbuffer? Do most people
enable it?
We have a PSTN server that has our RBS T1 trunks in a central
location,
then have clients that connect via SIP to us for access to those
trunks.
Most of them are
On Thu, 8 Apr 2010, Tim Nelson wrote:
- Jeff LaCoursiere j...@jeff.net wrote:
What is the consensus on using the 1.4 jitterbuffer? Do most people
enable it?
We have a PSTN server that has our RBS T1 trunks in a central
location,
then have clients that connect via SIP to us for
- Jeff LaCoursiere j...@jeff.net wrote:
On Thu, 8 Apr 2010, Tim Nelson wrote:
- Jeff LaCoursiere j...@jeff.net wrote:
What is the consensus on using the 1.4 jitterbuffer? Do most
people
enable it?
We have a PSTN server that has our RBS T1 trunks in a central
location,
Klaverstyn, David C wrote:
I have a Digium TE121 currently install in the server that the E1 ISDN
line is connected to. The Polycom has 4 by RJ45 connections for the
512kbit/s service.
It sounds like the Polycom device is expecting to be plugged into four
BRI (2B+D) lines, and then it will
Doug Lytle wrote:
Jeff LaCoursiere wrote:
On Thu, 8 Apr 2010, bruce bruce wrote:
Nope - unplugging a line that is in a hunt will result in Ring-No-Answer.
Ditto for previous advice to destroy the zap channel or to leave it out of
Our telecom guy said, that when you
Indeed the telco has no interest in changing the cable, and by the time they
send someone to look at the cable it's a sunny day and everything dried out.
Hence the order for PRI. Can't wait to fire it up tomorrow.
But, taking this number out of hunt is not so much of an option now as it
will cost
bruce bruce wrote:
Indeed the telco has no interest in changing the cable, and by the
time they send someone to look at the cable it's a sunny day and
everything dried out. Hence the order for PRI. Can't wait to fire it
up tomorrow.
Hope for your sake the same cable is not involved. Then
On Thu, 8 Apr 2010, John Novack wrote:
A simple short on the pair will fix that, though that would require you
to be on site, not always an option
Would sacrificing a spare line cord (cut, strip, twist together) be an
option for the on-site staff?
--
Thanks in advance,
Not really when you got call center people who deal with makeup goods :-)
and their manager can only break things. I can't trust them anywhere near
the server. Let alone me telling them which cable to short on the bix. I
would presist for Digium to come up with something that would allow soft
Hello.. maybe you can just have the telco do an immediate forward of that
number to the fifth number in the hunt group until it is fixed...
On Thu, Apr 8, 2010 at 1:15 PM, Steve Edwards asterisk@sedwards.comwrote:
On Thu, 8 Apr 2010, John Novack wrote:
A simple short on the pair will fix
On Thu, Apr 8, 2010 at 4:30 PM, David Backeberg dbackeb...@gmail.com wrote:
However, something is really weird when I need to do System() calls.
It almost feels like delay in reading loopback, or running out of
available files on the system, or something like that. I'm rebooted,
and the
Chris Miller wrote:
Understood, I figured it was something like that. Do you have some
mechanism in the source install that causes similar enforcement behavior?
No, because there's no practical way to do it. If someone downloads and
installs Asterisk and Asterisk-Addons from source, then
David Backeberg wrote:
I'm doing really, really innocent things, like:
exten = s,n,System(test -e ${MESSAGE_PATH}${EXTEN})
So I did some more testing. Same dialplan, reverted to
asterisk-1.6.0.13, and the contexts that do these test -e calls runs
lightning fast. It's like maybe there's
I've just upgraded to 1.6.2.6 on one of my test systems. I started out
happy, with some improvements in transfers to Local() channels from a
SIP channel, and much nicer verbose fax handling.
However, something is really weird when I need to do System() calls.
It was really, really weird. This was
Timothy C Litwiller wrote:
This upgrade says it has a special procedure and changes the layout of
the files it uses - so I am not sure I can downgrade again. I've asked
on the Pikawarp.org forum but so far no answer. if it goes a few more
days I will have to try something - the people in
On Thu, Apr 8, 2010 at 5:01 PM, Kevin P. Fleming kpflem...@digium.com wrote:
David Backeberg wrote:
I'm doing really, really innocent things, like:
exten = s,n,System(test -e ${MESSAGE_PATH}${EXTEN})
So I did some more testing. Same dialplan, reverted to
asterisk-1.6.0.13, and the contexts
Hi!
We are in process of setting up an audio guide that will cover notable places of
our capital Riga, Latvia.
The target audience are tourists that dials a free phone number from a mobile
handset to listen to a 3 minute introduction to historic place.
All audio, 10+ languages are recorder in
I am looking for something in asterisk that
will let me record a wav file in asterisk (which I know how to do)
then some other command (external or dialplan) that would read
the wave file and tell me if a certain tone or frequency is present.
Is this in asterisk already - any way to do it?
On Fri, 9 Apr 2010, Arkadi Shishlov wrote:
It would be essential to get your comments (in email or by leaving a
voice message) about sound quality if you could call the menu at
sip:1...@riga.beta.lv (actually, any number at riga.beta.lv)
I get:
-- Executing Dial(SIP/501-0961b3a8,
Hello All:
I saw there are app_fax and app_chanspy modules in 1.6.2.6, but there is NO
sample configure file for them.
Is anybody know how to use them, or where is the documentation for them?
Thanks
--
Refer to: http://www.microsuncn.com
Best Regards
Alan Zheng
--
I would not think you'd need to worry about jitter on a normal 100mbit
LAN unless there is heavy traffic or people are running their PC's through
the phone (don't remember if the 501 has two ethernet ports...). Typically
the quality issues are introduced on your WAN connectivity between the
Hi.
On the Spa 3102 is set as Dialplan s0:8028 on PSTN line tab, since other
way the incoming call will try to be routed to a non set extension on
[gw8028] context
Best Regards
Jose Flores Galicia
floj...@gmail.com
BriefCode Code Based Training
2010/4/8 Seann Clark nombran...@tsukinokage.net
Yes, the SPA-3201 is set as: (S0:8028) on dialplan 8, which is what I
have the device set to use. My bare bones working dialplan from
Callweaver works nearly perfectly with Asterisk, and takes all the calls
and works just as it did in Callweaver (making adjustments for the
differences in
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