Displaying 20 results from an estimated 8000 matches similar to: "did i miss the memo on asterisk devel ?"
2015 Jun 02
0
did i miss the memo on asterisk devel ?
sean darcy wrote:
> I usually lurk on the asterisk devel list to see what's going on.
>
> No posts for a week or two. Has the list moved ?
Nope - it's just been a quiet time.
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org
2018 Aug 30
3
Community forum ?
I see a lot of tag lines on posts for the Asterisk Community Forum. Is
that forum supposed to supersede this mailing list ?
sean
2014 Apr 26
2
asterisk servers down ?
I can't reach digium.com or asterisk.org. Did I miss the memo?
sean
2016 Mar 31
4
PJProject Bundled Update
As you know, the ability to use a bundled version of pjproject was
introduced with Asterisk 13.8.0.
More info on the Asterisk Wiki
<https://wiki.asterisk.org/wiki/display/AST/Building+and+Installing+pjproject#BuildingandInstallingpjproject-bundled>
and
in this email thread
<http://lists.digium.com/pipermail/asterisk-users/2016-March/288685.html>.
Since then I've fixed a few
2020 Sep 05
4
func_pjsip_aor.so: undefined symbol: ast_sip_location_retrieve_aor_contacts
asterisk-16.13.0-rc2. Fedora 32
pjsip won't load because of undefined symbols:
[Sep 4 14:19:25] ERROR[141137]: loader.c:2396 load_modules: Error
loading module 'func_pjsip_aor.so':
/usr/lib64/asterisk/modules/func_pjsip_aor.so: undefined symbol:
ast_sip_location_retrieve_aor_contacts
[Sep 4 14:19:25] ERROR[141137]: loader.c:2396 load_modules: Error
loading module
2018 Aug 30
3
Community forum ?
Is the list going to be the same after sangoma take over digium?
On Thu, Aug 30, 2018 at 11:12 AM, Joshua Colp <jcolp at digium.com> wrote:
> On Thu, Aug 30, 2018, at 12:05 PM, sean darcy wrote:
> > I see a lot of tag lines on posts for the Asterisk Community Forum. Is
> > that forum supposed to supersede this mailing list ?
>
> Both remain available but the community
2018 Aug 29
3
getting invites to rtp ports ??
I'm getting invites to very high ports every 30 seconds from a
particular ip address:
Retransmitting #10 (NAT) to 5.199.133.128:52734:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP
0.0.0.0:52734;branch=z9hG4bK1207255353;received=5.199.133.128;rport=52734
From: <sip:37120116780191250 at 67.80.191.250>;tag=1872048972
To: <sip:3712011972592181418 at 67.80.191.250>;tag=as3a52e748
2023 Nov 20
1
Finding old patches
Hi,
In the past when I wanted to back port a patch I would go on to the issue
tracker and find a link to the patches that were uploaded ( I think
through gerrit?). I am trying to see what changes were done for
https://issues-archive.asterisk.org/ASTERISK-26109. It seems the code
changes were introduced in 14.4.0-rc1. Is there any "n00b" way of seeing
what patches were created for this
2015 Aug 25
4
Ringback issue
My last problem was nicely solved through this mailing list so
hopefully this new problem will have the same happy outcome.
My situation is that I have many extensions. Here is a sample:
[client-phone](!)
type=friend
host=dynamic
secret=XXXXXXXXXX
dtmfmode=auto
disallow=all
allow=ulaw
allow=gsm
allow=g723
allow=ilbc
subscribemwi=no
[4165555555](client-phone)
secret=xxxxxxxxxxxxxxxxxxxxxxxxxx
2015 Aug 15
2
One way audio - doesn't seem to be NAT issue - SOLVED!
On Sat, 15 Aug 2015 16:30:39 +0800
Michael Dupree <michael at easybitllc.com> wrote:
> Not 100% ure, but maybe play with the canreinvite or directmedia
> settings.
Yes! That was it. Just for future searches here is what I did. I
added "directmedia = no" in sip.conf. This fixed the issue.
I believe that Asterisk was getting confused when one leg was inside
NAT and the
2019 Aug 22
2
h265 codec pass through on asterisk
Well, that sounds pretty straight forward.
I can do this and push it to gerrit.
Do I need to create a ticket for this?
With best regards
Florian Floimair
Innovation - Software-Development
COMMEND INTERNATIONAL GMBH
A-5020 Salzburg, Saalachstraße 51
http://www.commend.com <http://www.commend.com/>
Security and Communication by Commend
FN 178618z | LG Salzburg
Am 22.08.19, 11:55
2018 Apr 09
2
Asterisk behind NAT Early Media Video
My understanding based on Wireshark analysis is that the signaling works
(also the recipent phone is displaying the video frame before accepting the
call), also the calling phone send video (i see that also via Wireshark)
but the recipent phone doesn't get any video from the Asterisk before the
call.
2018-04-09 17:02 GMT+02:00 Joshua Colp <jcolp at digium.com>:
> On Mon, Apr 9,
2019 Apr 08
2
pjsip endoint woes
On Sat, Apr 6, 2019, at 10:04 AM, sean darcy wrote:
> On 4/5/19 10:36 AM, sean darcy wrote:
> > I'm trying to set up pjsip to work with an obi202 and google voice. But
> > I can't configure the endpoint.
> >
> > pjsip:
> >
> > [obi202-auth](!)
> > type = auth
> > auth_type = userpass
> > password = <mypass>
> >
>
2019 Aug 22
2
h265 codec pass through on asterisk
All,
I'm using asterisk 16.4.0 with h264 and opus quite well using linphone 4.1
client on android and baresip on linux.
I'm exploring use of h265 for improved video quality/lower network
bandwidth. I do not see pass through support on asterisk for h265/hvec. All
my SIP clients and underlying hardware have hvec/h265 encoding and decoding
available.
I would have liked vp9 however, vp9
2014 Dec 21
2
11.5.0: blindxfer problems [Spam score:10%]
Have you enabled DTMF logging and seen the DTMF codes being recognised by
Asterisk? I had a bunch of soft phones that I had to change to using ?sip
info? for the DTMF signalling as the RFC signalling was not always being
recognised. This would cause transfers to appear as if the user had not
dialled any digits.
On 20/12/2014 20:52, "sean darcy" <seandarcy2 at gmail.com> wrote:
2010 Jun 18
6
Why asterisk down when inet server down?
We have a 10.10.0.0 internal network. The asterisk server - 10.10.10.180
- has a PRI connection to a T-1. Another server is the router to the
internet. All phones in the office and the workstations are on the network.
Most of the internal phones are aastra 9133i's. Here the network config
from a phone:
Network Settings
Basic Network Settings
DHCP [ ]
2013 Mar 07
2
11.3: how to hang up on google voice
Some calls I get from google voice, I just send myself an email about
the call and want to hangup. But I can't seem to make gv know I've hung up.
extensions.conf:
same => n,GoToIf($["${CALLERID(num)}"="office"]?email)
.................
same => n(email),System(/usr/local/bin/emailme........)
same => n,Answer() ; also tried without this
same =>
2018 Aug 29
2
getting invites to rtp ports ??
On 08/29/2018 09:42 AM, Carlos Rojas wrote:
> Hi
>
> Probably somebody is trying to hack your system, you should block that
> ip on your firewall.
>
> Regards
>
> On Wed, Aug 29, 2018 at 9:34 AM, sean darcy <seandarcy2 at gmail.com
> <mailto:seandarcy2 at gmail.com>> wrote:
>
> I'm getting invites to very high ports every 30 seconds from a
2018 Aug 29
3
getting invites to rtp ports ??
On 08/29/2018 11:59 AM, Telium Support Group wrote:
> Block a single IP is the wrong approach (whack-a-mole). You should consider a more comprehensive approach to securing your VoIP environment. Have a look at this wiki:
>
> https://www.voip-info.org/asterisk-security/
>
>
>
> -----Original Message-----
> From: asterisk-users [mailto:asterisk-users-bounces at
2011 Dec 26
5
how to listen on different sip port for a device?
I'm trying to allow access to the office from home. But the ip provider
(cablevision) blocks udp 5060. I can see the register packets leaving on
wireshark, but nothing received by office. Changed to port to 6111 and
now the packets show up.
In the server I've set port=6111 in the device in sip.conf, but * is NOT
listening for 6111:
netstat -an | grep 5060
tcp 0 0