Hi all.
Just wanted to let people know about a small project I started over the weekend
to help me keep up with news about Asterisk. http://asterisktimes.xdev.net/
Some of the other new sites are either not there anymore or slow to update, so
I've come up with a different idea for keeping
If I understand correctly, setting
encryption=no
means that Asterisk will make outgoing calls without encryption, but
will be happy to accept incoming calls regardless of whether the caller
wants encryption or not
If encryption=yes, then Asterisk not only uses encryption for the
outgoing calls
On Tue, Jan 28, 2014 at 2:40 AM, Ira i...@extrasensory.com wrote:
Hello Matthew,
Monday, January 27, 2014, 1:49:44 PM, you wrote:
Do you have the exact error message that pjproject gave when you ran
into this problem?
I don't, but I guess I can reinstall the offending software to get
I have been trying to get a feel for scaling or dimensioning using asterisk
11.
if I desire to use something like a dell r320, hardware RAID, 2G E5-2420,
4G RAM
and only SIP trunking using gsm (least bandwidth and no transcoding)
how many calls out can I expect to make at one time and asterisk
On 28/01/14 15:01, Jerry Geis wrote:
I have been trying to get a feel for scaling or dimensioning using
asterisk 11.
if I desire to use something like a dell r320, hardware RAID, 2G
E5-2420, 4G RAM
and only SIP trunking using gsm (least bandwidth and no transcoding)
how many calls out can I
On Thu, Jan 23, 2014 at 9:05 PM, Daniel Jenkins dan.jenkin...@gmail.comwrote:
On Thu, Jan 23, 2014 at 8:46 PM, Matthew Jordan mjor...@digium.comwrote:
On Thu, Jan 23, 2014 at 9:31 AM, Michelle Dupuis mdup...@ocg.ca wrote:
Thanks - I've been through that doc before and couldn't find the
Hello All,
I've asked this on the asterisk-dev list, so sorry for cross-posting. So
far I'm not sure how to accomplish this without looking at the source code
or looking at some other way to get around this issue.
I'm trying to have an automated call to an Aastra SIP phone and have the
call
On Tue, Jan 28, 2014 at 6:29 AM, Matthew Jordan mjor...@digium.com wrote:
On Tue, Jan 28, 2014 at 2:40 AM, Ira i...@extrasensory.com wrote:
Hello Matthew,
Monday, January 27, 2014, 1:49:44 PM, you wrote:
Do you have the exact error message that pjproject gave when you ran
into this
On 28/01/14 16:56, Steve McCann wrote:
Hello All,
I've asked this on the asterisk-dev list, so sorry for cross-posting.
So far I'm not sure how to accomplish this without looking at the
source code or looking at some other way to get around this issue.
I'm trying to have an automated call
On Tue, Jan 28, 2014 at 10:56 AM, Steve McCann srmcc...@gmail.com wrote:
Hello All,
I've asked this on the asterisk-dev list, so sorry for cross-posting. So far
I'm not sure how to accomplish this without looking at the source code or
looking at some other way to get around this issue.
I'm
Although I haven't tried this for this particular example, instead of
using a .call file, you could probably originate a call using Ryan Bullock's
Asterisk::AMI PERL module
http://search.cpan.org/~greenbean/Asterisk-AMI-v0.2.8. It's one of the most
valuable tools that I have and I've written
Hello,
I'm not sure if my previous e-mail went through, it contained a link so it
may have been blocked, Apologies if this is a duplicate.
I have a card here at the office that I'm trying to validate with some
tests. For now I've followed the instructions available in the KB
Hi all,
I'm having issues with overwrite caller id, when I call someone my caller
id should be mycompanyinc but instead my id shows up as my extension
number 101.
this is what i have in sip.conf
[101]
type=friend
context=sipphones
call-limit=99
callerid=iuser 101
disallow=all
allow=ulaw
Hello;
Is there a method way to be able to dial the phone number through asterisk
from the outlook email contact?
Regards
Bilal--
_
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New to Asterisk? Join us
Hi
Yes, there is, I am using
http://outcall.sourceforge.net/
it's opensource.
On Tue, Jan 28, 2014 at 2:13 PM, bilal ghayyad bilmar...@yahoo.com wrote:
Hello;
Is there a method way to be able to dial the phone number through
asterisk from the outlook email contact?
Regards
Bilal
On 1/28/14, 1:55 PM, motty cruz wrote:
Hi all,
I'm having issues with overwrite caller id, when I call someone my
caller id should be mycompanyinc but instead my id shows up as my
extension number 101.
this is what i have in sip.conf
[101]
type=friend
context=sipphones
call-limit=99
We have used this commercial software to dial via our IP phones at my
office. It's about $10 a license IIRC.
http://www.theteletrigger.com/
On Tue, Jan 28, 2014 at 12:13 PM, bilal ghayyad bilmar...@yahoo.com wrote:
Hello;
Is there a method way to be able to dial the phone number through
On Tue, 28 Jan 2014 10:55:58 -0800
motty cruz motty.c...@gmail.com wrote:
this is what i have in extensions.conf
[outbound]
exten = _91NXXNXX,1,Set(CALLERID(num)=mycompanyinc)
exten = _91NXXNXX,2,Dial(SIP/att/${EXTEN:1},80)
exten = _9NXX,1,Set(CALLERID(num)=mycompanyinc)
exten =
Thank you for your reply, I updated extensions.conf file to reflect your
suggestion, I will monitor Asterisk for any more issues,
Thanks,
On Tue, Jan 28, 2014 at 11:23 AM, Andres and...@telesip.net wrote:
On 1/28/14, 1:55 PM, motty cruz wrote:
Hi all,
I'm having issues with overwrite
Bonjour,
Depuis que j'ai passé une machine de Squeeze à Wheezy, j'ai toutes les
secondes dans /var/log/syslog, ce message :
udevd[374]: timeout: killing '/sbin/modprobe -b
There's also SIPTAPI: http://www.ipcom.at/en/telephony/siptapi/
On Tue, Jan 28, 2014 at 7:13 PM, bilal ghayyad bilmar...@yahoo.com wrote:
Hello;
Is there a method way to be able to dial the phone number through
asterisk from the outlook email contact?
Regards
Bilal
--
Wrong list, wrong language :
double appologize for my previous message ..
Regards
--
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-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
All;
I'm working on a project (using Asterisk 1.8, but 11 would probably work
just as well) where so far I've been able to originate over 1,000 concurrent
outbound faxes. I have no problem with that so far. Where I have the problem
is that Asterisk is dumping core after the faxes are sent.
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