Re: [asterisk-users] OT - Merry Christmas and a Happy and Prosperous 2014
To you as well. On Dec 25, 2013 8:56 AM, Nick Cameo sym...@gmail.com wrote: God Bless and Merry Christmas to All! Nick. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Pulse Audio Motorboating Audio with Asterisk
https://bbs.archlinux.org/viewtopic.php?pid=920549
On Sat, Jun 8, 2013 at 9:51 AM, Jerry Geis ge...@pagestation.com wrote:
When I use pulse audio and Asterisk 11.4.0 on the Console/Dsp port
I get a motorboating sound or warble - or - just not clear audio.
When I switch that to ALSA direct it sounds just fine.
What might be happening with pulse audio that it does not
sound clear???
asound.conf below.
Thanks,
Jerry
more /etc/asound.conf
#
# Place your global alsa-lib configuration here...
#
@hooks [
{
func load
files [
/etc/alsa/pulse-default.conf
]
errors false
}
]
--
__**__**_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/**mailman/listinfo/asterisk-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users
--
Rob Krakora
MessageNet Systems
101 East Carmel Dr. Suite 105
Carmel, IN 46032
(317)566-1677 Ext 212
(317)663-0808 Fax
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Pulse Audio Motorboating Audio with Asterisk
Pulse Audio 4.0 just came out and has gotten good reviews as it improves
audio quality...I installed it on the devel and support mediaports and will
test tomorrow.
http://www.freedesktop.org/wiki/Software/PulseAudio/Notes/4.0/
On Mon, Jun 10, 2013 at 7:59 PM, Robert Krakora
rob.krak...@messagenetsystems.com wrote:
https://bbs.archlinux.org/viewtopic.php?pid=920549
On Sat, Jun 8, 2013 at 9:51 AM, Jerry Geis ge...@pagestation.com wrote:
When I use pulse audio and Asterisk 11.4.0 on the Console/Dsp port
I get a motorboating sound or warble - or - just not clear audio.
When I switch that to ALSA direct it sounds just fine.
What might be happening with pulse audio that it does not
sound clear???
asound.conf below.
Thanks,
Jerry
more /etc/asound.conf
#
# Place your global alsa-lib configuration here...
#
@hooks [
{
func load
files [
/etc/alsa/pulse-default.conf
]
errors false
}
]
--
__**__**_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/**mailman/listinfo/asterisk-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users
--
Rob Krakora
MessageNet Systems
101 East Carmel Dr. Suite 105
Carmel, IN 46032
(317)566-1677 Ext 212
(317)663-0808 Fax
--
Rob Krakora
MessageNet Systems
101 East Carmel Dr. Suite 105
Carmel, IN 46032
(317)566-1677 Ext 212
(317)663-0808 Fax
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk disconnecting SIP Calls after 15 Minutes
I am having the same problem with Asterisk 11.2.0 and Linphone and it is exactly 15 minutes and occurring with SIP running on our LAN. On Thu, Mar 21, 2013 at 3:31 AM, Florian Wolters flor...@florian-wolters.de wrote: Hi @ll, I just moved my Asterisk Box and changed the Provider and Internet Access to a full IP Access by Deutsche Telekom. I set up my sip.conf as I found various examples throughout the Net. Calls and some other stuff is basically working. The problem I ran into is, that the outgoing and incoming calls are dropped after exactly 15 Minutes. Solution for this should be setting the session-timers to refuse but this doesnt change anything here. I am running Asterisk 1.6 on Ubuntu 12.04 LTS and also tried the latest Asterisk by Digium without success. Has anyone else has the Same problem or is a solution already known? Could someone point me in the right direction? I can provide (debug) logs if essential. Best regards Flo -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Rob Krakora MessageNet Systems 101 East Carmel Dr. Suite 105 Carmel, IN 46032 (317)566-1677 Ext 212 (317)663-0808 Fax -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] app_rtsp.c ported to Asterisk 11.x
Hi, If anyone is interested, I have ported app_rtsp.c to Asterisk 11.x. I have tested it with GStreamer RTSP server and a C920 webcam streaming H264 SVC video from one machine to another machine running Linphone. Contact me at this e-mail address robkrak...@messagenetsystems.com for source code. Best Regards, -- Rob Krakora MessageNet Systems 101 East Carmel Dr. Suite 105 Carmel, IN 46032 (317)566-1677 Ext 212 (317)663-0808 Fax -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
