05.03.2015 11:29, Dmitry Melekhov пишет:
Hello!
Just installed asterisk 13.2.0 and see many such messages in log, I
see them in console during calls, really something like this:
-- Executing [6166@kanbaikal:2] Dial("OOH323/kanbaikal-6",
"SIP/6166@asterisk") in new stack
== Using SIP
Hello!
Just installed asterisk 13.2.0 and see many such messages in log, I see
them in console during calls, really something like this:
-- Executing [6166@kanbaikal:2] Dial("OOH323/kanbaikal-6",
"SIP/6166@asterisk") in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS ma
I am trying to determine how the transfer button on the Snom-870 works
with Asterisk. Is the ## special code employed as when it is entered
through the handset or is the blind transfer through the phone
function accomplished in a different fashion?
--
*** E-Mail is NOT a SECURE channel
Joshua Colp wrote:
> Chirag Desai wrote:
> >* Joshua Colp wrote:
*> >> >*
*> >> >* > Remove "transport=transport-tcp" from your endpoints.
*> >> >> >* Joshua...I did that but now my endpoints won't register.
*
> That should have no impact on things. Can you clarify what you mean by
> it doesn't
Chirag Desai wrote:
Joshua Colp wrote:
> Remove "transport=transport-tcp" from your endpoints.
Joshua...I did that but now my endpoints won't register.
That should have no impact on things. Can you clarify what you mean by
it doesn't register? What happens?
--
Joshua Colp
Digium, Inc.
Joshua Colp wrote:
> Remove "transport=transport-tcp" from your endpoints.
Joshua...I did that but now my endpoints won't register.
Kind Regards,
Chirag
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Chirag Desai wrote:
My endpoint looks like this:
[user1]
type=endpoint
transport=transport-tcp
context=local_out
disallow=all
allow=alaw
allow=ulaw
allow=g722
auth=user1
aors=user1
direct_media=no
rtp_symmetric=yes
force_rport=yes
rewrite_contact=yes
Remove "transport=transport-tcp" from yo
Hi all,
I have Asterisk 13 running and I'm currently trying to get PJSIP working on
TCP.
My transport looks like this. My box is not behind NAT.
[transport-tcp]
type=transport
protocol=tcp
bind=0.0.0.0:5061
My endpoint looks like this:
[user1]
type=endpoint
transport=transport-tcp
context=loca
Sorry, i found the source of problem.
https://wiki.asterisk.org/wiki/display/AST/Dialing+PJSIP+Channels
dialing via pjsip have to change dialplan syntax :(
May be it will be usefull sombody else.
04.03.2015 21:54, Dmitriy Serov пишет:
Hello.
I am using asterisk and chan_sip a lot of years. And
All;
I build a conference server using Asterisk 1.8 and the third party
module app_konference.so. I would ask on their forum, but the forum seems to
be pretty dead. The problem I am having is that when I have conferences that
have a lot of members, say 100+ users, the DTMF seems to not work. F
Im facing some problems with RTP during queue agent calls.
Randomly during the call the agent can't hear the other side. This happens
for two or three seconds and the the call continue without problems.
The weird thing is that the recording for this call is fine, so both sides
are recorded withou
For the mailing list archive and for anyone else interested.
A few years ago we needed to automatically run a second AGI if the first AGI
failed i.e. a "failsafe" setup. Mainly because I'm not a very good programmer.
8-|
The code below is very similar to what we use in production. This code
I'll also warn that if you do intend on doing anything with
rtpengine/webrtc2sip etc. you'll need Openssl 1.0.1j or better so that it
has the proper support for DTLS-SRTP. This means you are sort of SOL if you
are running CentOS5 unless you plan on building Openssl 1.0.1 manually. The
websocket mod
Hello.
I am using asterisk and chan_sip a lot of years. And newbie in chan_pjsip.
Now i am transfering all from chan_sip to chan_pjsip. And have a lot of
questions. First of...
system: Asterisk 13.2 on slackware 14.1
Errors on outgoing call:
[2015-03-03 00:18:58] ERROR[6794]: chan_pjsip.c:1778
On Wed, Mar 4, 2015 at 12:47 AM, Jarrod Cuzens wrote:
> For those that were interested I have attached the kamailio.cfg which we
> have working with Kamailio 4.2.1 and Asterisk 1.8.23/32. Specifically, the
> following yum packages:
>
> kamailio.x86_64 4.2.1-4.1
> @home_kamaili
I'm having some problems with a vega sangoma, if a call comes into my
ivr and hangs up, the call continues to ring and leaves hanging the
channel, I have to restart Asterisk and everything works Ok
my sangoma is a vega 50 , 4 FXO .
I tried different tone of countries and does not work,
this is t
This seems to me to be getting down to some sort of problem with
configuring the Snom-870.
when I register the device 41712 (set up for transport=tls only) then
I see this in the SIP trace:
Sent to udp:192.168.6.9:5060 at 4/3/2015 09:07:36:813 (836 bytes):
REGISTER sip:voinet09.internal.hamilto
Stuck with TLS transport,
I have 2 phones (both in local network for tests)
one connected with up second with tls
when I calling TLS to UPD everything is fine, but when UDP calls TLS I getting
an error
ERROR[44230]: pjsip:0 : tlsc0x7f143012 TLS connect() error: Connection
refused [code=1
If you would like to set things up via the GUI on your incredible PBX, you
could use queues instead of call groups (making your SIP clients agents of
the appropriate queues), and in the queues configuration page there is an CID
Name Prefix option, which allows you to add a label that will show up a
Background: I dabbled with asterisk years ago, and more recently have
more-or-less functioning IncrediblePBX systems for experimenting, but
I want to understand more so I'm now working with distro packages
(Ubuntu) and hand edited configurations files.
I have three SIP "trunks", each providing me
Markos Vakondios gmail.com> writes:
>
> Hi,
> I use a simple scheme:
>
> SIP video phone A (h264/Asterisk 1.8.11) <---IAX2 trunk> SIP video
phone B (h264/Asterisk 11.7.0)
>
> When calls from A to B and vice versa drop on pickup.
>
> On B side:
>
>
> [Oct 24 16:33:49] DEBUG[15590][C-000
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