On 20 March 2014 19:24, Dan Cropp d...@amtelco.com wrote:
Anyone know of a tutorial for configuring WebRTC on Asterisk 12 using
PJSIP?
Some useful stuff here, it's video's from last Astricon:
https://www.youtube.com/playlist?list=PLighc-2vlRgSwgJCxEh6NZwC8lE6XogaP
This particular session
meanwhile i got an answer from fprior who was testing patch already.
add this code to chan_sip.c :
/* Allow domain to be overridden */
if (!ast_strlen_zero(p-fromdomain))
d = p-fromdomain;
else /* Save for any further attempts */
I'm trying to determine the capacity of my host running Asterisk 11.8.1 on
CentOS 6.5.
The host is an Intel E3-1240v3 with 8GB RAM, an SSD, and gigabit Ethernet.
The primary application will be bridging groups of users using meetme().
I'm using 2 boxes -- 1 to initiate calls using call files
On Fri, 21 Mar 2014, Steve Edwards wrote:
The call file on box1 originates a call to box2 and then plays a 2 hour
WAV file.
The call file on box1 originates a SIP call to box2 and then plays a 2
hour WAV file.
--
Thanks in advance,
Built asterisk 11.8.1 on a Debian VPS. Testing using ekiga.
If h261 is checked in ekiga's video format list I have video, and
mouse over the video window shows it to be using h261.
But then I get the following lines a dozen or more times in the CLI:
[Mar 21 16:25:32]
If h261 is checked in ekiga's video format list I have video, and
[Mar 21 16:25:32] WARNING[31818][C-0010]: file.c:1241
ast_writefile: No such format 'h261'
Ekiga can do SIP. Maybe try that? And set/prioritize the codec in
ekiga to desired codec, not h261.
--
On (21/03/14 13:28), Adrian Serafini wealwild...@wombit.com put forth the
proposition:
If h261 is checked in ekiga's video format list I have video, and
[Mar 21 16:25:32] WARNING[31818][C-0010]: file.c:1241
ast_writefile: No such format 'h261'
Ekiga can do SIP. Maybe try that? And
H.323 is a communications protocol like SIP. H261 is a codec like ulaw or
gsm. You do not need H323 unless you are using the H323 protocol INSTEAD
of SIP.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf
I found below here: http://www.voip-info.org/wiki/view/Asterisk+cmd+MeetMe
If you have too many conferences, one CPU may not be able to mix all the
audio and you will have audio problems even if there are 7+ other CPUs that
are essentially idle while waiting for one CPU to mix everything. You
On (21/03/14 13:54), Steve Totaro stot...@totarotechnologies.com put forth
the proposition:
I found below here: http://www.voip-info.org/wiki/view/Asterisk+cmd+MeetMe
If you have too many conferences, one CPU may not be able to mix all the
audio and you will have audio problems even if there
H.323 is a communications protocol like SIP. H261 is a codec like ulaw or
gsm. You do not need H323 unless you are using the H323 protocol INSTEAD
of SIP.
I see. In Ekiga video codec window they are listed like:
[ ] h26190kHz H.323. SIP
etc.
Which is what I'm going by. The
On Fri, 21 Mar 2014, Steve Totaro wrote:
I found below here:
http://www.voip-info.org/wiki/view/Asterisk+cmd+MeetMe
If you have too many conferences, one CPU may not be able to mix all the
audio and you will have audio problems even if there are 7+ other CPUs
that are essentially idle
Coincidentally, 512 is my target. Any clues on how to get 200 more?
Upgrade to 1.4? hehe, I thought you were the self proclaimed 1.2
luddite? I'm a big fan of older releases with 1 year plus of uptime.
--
_
--
On 03/21/2014 02:09 PM, David Woodfall wrote:
H.323 is a communications protocol like SIP. H261 is a codec like
ulaw or gsm. You do not need H323 unless you are using the H323
protocol INSTEAD of SIP.
I see. In Ekiga video codec window they are listed like:
[ ] h26190kHz H.323. SIP
We noticed issues with voicemail and somehow looks like voicemail.conf has
been overwritten:
;!
;! Automatically generated configuration file
;! Filename: voicemail.conf (/etc/asterisk/voicemail.conf)
;! Generator: AppVoicemail
;! Creation Date: Thu Mar 20 06:48:16 2014
;!
i saw a bug for 1.4
On Fri, Mar 21, 2014 at 2:26 PM, Steve Edwards asterisk@sedwards.comwrote:
On Fri, 21 Mar 2014, Steve Totaro wrote:
I found below here: http://www.voip-info.org/
wiki/view/Asterisk+cmd+MeetMe
If you have too many conferences, one CPU may not be able to mix all the
audio and you will
On Fri, 21 Mar 2014, Adrian Serafini wrote:
Upgrade to 1.4? hehe, I thought you were the self proclaimed 1.2
luddite? I'm a big fan of older releases with 1 year plus of uptime.
Yep, that's me :)
I'm trying to make the leap from 1.2 to 11.8.1
--
Thanks in advance,
Steve Edwards wrote:
On Fri, 21 Mar 2014, Adrian Serafini wrote:
Upgrade to 1.4? hehe, I thought you were the self proclaimed 1.2 luddite? I'm
a big fan of older releases with 1 year plus of uptime.
Yep, that's me :)
I'm trying to make the leap from 1.2 to 11.8.1
That is a HUGE leap
On (21/03/14 15:20), Adrian Serafini wealwild...@wombit.com put forth the
proposition:
On 03/21/2014 02:09 PM, David Woodfall wrote:
H.323 is a communications protocol like SIP. H261 is a codec like
ulaw or gsm. You do not need H323 unless you are using the H323
protocol INSTEAD of SIP.
On (21/03/14 20:07), Dave Woodfall d...@dawoodfall.net put forth the
proposition:
On (21/03/14 15:20), Adrian Serafini wealwild...@wombit.com put forth the
proposition:
On 03/21/2014 02:09 PM, David Woodfall wrote:
H.323 is a communications protocol like SIP. H261 is a codec like
ulaw or
On (21/03/14 15:20), Adrian Serafini wealwild...@wombit.com put forth the
proposition:
On 03/21/2014 02:09 PM, David Woodfall wrote:
H.323 is a communications protocol like SIP. H261 is a codec like
ulaw or gsm. You do not need H323 unless you are using the H323
protocol INSTEAD of SIP.
On Fri, Mar 21, 2014 at 3:22 PM, Al lists asteris...@gmail.com wrote:
We noticed issues with voicemail and somehow looks like voicemail.conf has
been overwritten:
;!
;! Automatically generated configuration file
;! Filename: voicemail.conf (/etc/asterisk/voicemail.conf)
;! Generator:
On Fri, Mar 21, 2014 at 11:53 AM, Steve Edwards
asterisk@sedwards.com wrote:
I'm trying to determine the capacity of my host running Asterisk 11.8.1 on
CentOS 6.5.
The host is an Intel E3-1240v3 with 8GB RAM, an SSD, and gigabit Ethernet.
The primary application will be bridging groups
On Fri, 21 Mar 2014, Paul Belanger wrote:
DAHDI has a pseudo channel limit of 512, somebody has already posted how
to change it with modprode.
Not in this thread, but big thanks for the clue. Googling 'dahdi pseudo
channel limit modprobe' showed the secret sauce.
I can get 1,000
On Fri, 21 Mar 2014, Steve Edwards wrote:
Googling 'dahdi pseudo channel limit modprobe' showed the secret sauce.
Oops. Guess I should complete the thread...
You can set the DAHDI pseudo channel limit in /etc/modules.conf:
options dahdi max_pseudo_channels=x
or you can set it from
Is there any good documentation on that process?
On Fri, Mar 21, 2014 at 3:36 PM, John Novack
jnov...@stromberg-carlson.orgwrote:
Steve Edwards wrote:
On Fri, 21 Mar 2014, Adrian Serafini wrote:
Upgrade to 1.4? hehe, I thought you were the self proclaimed 1.2
luddite? I'm a big fan of
On Fri, 21 Mar 2014, Steve Edwards wrote:
I'm trying to make the leap from 1.2 to 11.8.1
On Fri, 21 Mar 2014, Steve Totaro wrote:
Is there any good documentation on that process?
I haven't looked. I know they added a few of variables to the AGI
environment Asterisk passes to your AGI on
In no specific order:
Download the Asterisk tarball you want to use and study all the
UPGRADE*.txt files included in it.
Buy or download the latest ATFOT book, study it.
Install Asterisk into a test box, even a VM is OK for testing, study the
output of core show applications and
passwordlocatio seems to be related to vmsecret
from voicemail.conf sample :
; passwordlocation=spooldir
; Usually the voicemail password (vmsecret) is stored in
; this configuration file. By setting this option you
can
; specify where
looking more into this, looks like this is not a issue, its related to
users changing voicemail password from handset, asterisk rewrites the file.
On Fri, Mar 21, 2014 at 9:31 PM, Al lists asteris...@gmail.com wrote:
passwordlocatio seems to be related to vmsecret
from voicemail.conf sample
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