and unavailable.
--
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
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and offline.
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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
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,
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Joshua Colp
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Check us out at: www.digium.com www.asterisk.org
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on now,
then I 'd be happy to help out with porting and testing.
There is one review at https://reviewboard.asterisk.org/r/3488/ which is
a proof of concept, but I know of noone actively working on it.
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville
work which
forwards logging messages from the PJ core into Asterisk log messages.
--
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
Daniel Pocock wrote:
On 22/07/14 18:20, Joshua Colp wrote:
Daniel Pocock wrote:
snip
FYI, I'm using the Debian packages, latest is 11.10.2~dfsg-1~bpo70+1
Has the chan_motif / xmpp / ICE stuff changed significantly in 12.x
releases?
Nope.
Is there any way I can enable ICE debugging
then it will not be associated with the account and the
Jingle support will not be present.
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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
to do the
matching based on the source IP address OR use a user account with
authentication. If using the user account the user portion of the From
header has to be set to the username (from_user in pjsip, fromuser in
chan_sip).
Cheers,
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445
worked on it.
Cheers,
--
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
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are calling
who) and how you want it to work I can't answer.
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Joshua Colp
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Check us out at: www.digium.com www.asterisk.org
try using the VOLUME dialplan function to increase the volume
manually, but this will impact what you hear as well.
Cheers,
--
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
of the call itself. This can be accomplished
using SIP session timers. There is a section SIP Session-Timers in the
sip.conf.sample file which has the various configuration options
relating to it.
Cheers,
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Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW
it
was not possible to monitor the signaling of the call and is an
Asterisk-ism. You've got a few options, though:
1. Increase the rtpholdtimeout
2. Don't use rtpholdtimeout and use SIP session timers instead (check
the SIP Session-Timers section in sip.conf.sample)
Cheers,
--
Joshua Colp
Digium
to these REGISTERs Kamailio sees 2 AORs for the
account for those clients whose 'Subscribe to MWI' setting is defined as
'both'.
Can you provide a link to a sip debug log of this occurring? It sounds
extremely weird and I'm not really sure how chan_sip would be doing such
a thing...
Cheers,
--
Joshua Colp
peers?
--
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
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= username:secret@host/callbackextension
;
; and more readable because you don't have to write the parameters in
two places
; (note that the port is ignored - this is a bug that should be fixed).
Remove that column from your table, restart Asterisk, and it should go away.
--
Joshua Colp
Digium
' but ICE support not available
-- Executing [s@xmpp-in:1] NoOp(Motif/allan-ce76, llamada de
usuario XMPP ) in new stack
Do you have the uuid development library installed? It is an optional
dependency and without it res_rtp_asterisk will not be built with ICE
support.
--
Joshua Colp
Digium
Justin Killen wrote:
Is there a channel variable / status indicator / function that indicates
if the current channel has been answer()’ed?
${CHANNEL(state)} will return the state the channel is currently in. If
the channel is answered the state will be Up.
Cheers,
--
Joshua Colp
Digium
dirtectmedia=yes
directsetup=yes
I am using asterisk version 12.3
Remove the nat option. What does the console output show when making a
call between two SIP devices?
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Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out
is expecting. If they don't
match, then no go. One way to achieve this could be a view in your database.
Cheers,
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Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com www.asterisk.org
is reproducible every time.
Our asterisk box is behind nat.
Please Help,
If you can provide console output with SIP traces as well as rtp debug
then someone may be able to help. Without this information there's not
really any suggestion or comment I can offer.
Cheers,
--
Joshua Colp
Digium, Inc
logic (such as the Dial above). That should do what you want.
Cheers,
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com www.asterisk.org
to
192.168.1.191:8000 http://192.168.1.191:8000
== WebSocket connection from '192.168.1.191:54390
http://192.168.1.191:54390' closed
Are either side using encryption or ICE?
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out
Sameer Rathod wrote:
yes I had configured
icesupport=yes ;
Asterisk does not support direct media establishment (with either
chan_sip or chan_pjsip) if secure media (SRTP) or ICE is in use.
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL
will therefore have to explicitly specify it:
Dial(PJSIP/default_outbound_endpoint/sip:${EXTEN}@10.10.10.2)
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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
option to set the subscription
context the context specified by context is used.
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Joshua Colp
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445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com www.asterisk.org
and 6002 respectively.
What am I missing? It was working in sip and not in pjsip.
The underlying event package (dialog-info+xml) used by Grandstream and a
few other vendors is not currently supported in the PJSIP module.
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Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW
here responses and in console No compatible
codecs, not accepting this offer
No, Asterisk does not allow passing arbitrary stuff like this.
Cheers,
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Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com
of a format is not currently something that can be done just
by loading a module.
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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
to change this so that reloading Asterisk after a
dialplan change affects new calls and not calls in progress?
Not without drastically changing the PBX core.
Cheers,
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Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
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Fronc Hias wrote:
Hi!
sorry to poke in... but i haven't heard anything since posting my logs :(
No real additional thoughts. Everything looks as though it should work.
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Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
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CDR wrote:
When a client send me an INVITE with this type of caller ID
From:
eurussip:null@XX.XX.XX.XXX;tag=3430296121-3809549020-352327076-1077499159
Asterisk 14 sends back
SIP/2.0 500 Server error occurred (1/SL)
This response is not from Asterisk, it's from an OpenSIPS server.
--
Joshua
output
with debug and create an issue[1] as this sounds like a bug.
Cheers,
[1] https://issues.asterisk.org/jira
--
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
Fronc Hias wrote:
FYI: Joshua Colp already replied to my initial post of this message in
asterisk-app-dev.
he suggested to move it here (asterisk-users)
he so far stated, that early media/Video should theoretically work...
but probably no one tried this in recent times...
looking foreward
though where it is 8:09AM. Not t early.
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Joshua Colp
Digium, Inc. | Senior Software Developer
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Check us out at: www.digium.com www.asterisk.org
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will happen. Just so
we're on the same page here...
#1 Do you mean that you make a change, do a reload, and nothing happens.
OR
#2 That you make a change, do two reloads, and the second one does nothing.
I was under the impression that #1 was going on.
--
Joshua Colp
Digium, Inc. | Senior Software
Sean Darcy wrote:
I can't reach digium.com or asterisk.org. Did I miss the memo?
I have opened a ticket with IT. I'll keep the list apprised when the
problem is isolated and resolved.
Cheers,
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL
William Hetherington wrote:
Both resolve fine for me :)
It seems to be sporadic. I suspect one of the DNS servers is having
issues, so it resolves fine for some and gets cached - for others not so
much.
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Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW
come off as harsh; I do not mean it to be
so, since it is what I want. :)
Yes.
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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
debugging, but will doing so also
block secure media?
The media is not carried over the SIP signaling, it is negotiated using
SDP and flows over different ports. Unless you also do media
manipulation in the SIP proxy then it won't touch that.
--
Joshua Colp
Digium, Inc. | Senior Software
?
Asterisk does not currently support this configuration (specifically
passing through candidates and ICE information like this).
Cheers,
--
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
a device may have registered with to an AOR. As
you've figured out using fromuser does allow the user endpoint
identifier to find the endpoint and then it works happily.
Cheers,
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us
. If playing back
files you've got disk I/O going on which can slow stuff down (unless the
system caches it enough or you throw them into a ramdisk yourself). You
are also sending and receiving a *ton* of small packets. This can make
network equipment and NICs unhappy.
Cheers,
--
Joshua Colp
Digium
it to work. This makes
documentation as close to a first class citizen as it can be (albeit
that doesn't mean one can't write bad documentation).
Cheers,
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com
, but nothing existing has been migrated
to it or even will be (it's a fundamentally shift).
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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
Bruce Ferrell wrote:
On 04/17/2014 01:34 PM, Joshua Colp wrote:
Bruce Ferrell wrote:
I was just told that realtime was no longer in asterisk 12, but I find
enhancements in 12.2-rc2 and no sign in the wiki that this is true.
Can someone comment?
Realtime has not been removed or deprecated
peer found
Your client is configured to use 6004 as the username while your above
configuration uses Peter. Since the usernames differ, it fails.
Cheers,
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Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com
when doing development and we try our hardest to
make the experience of upgrading as painless as possible from that
perspective.
Cheers,
--
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
you will need to escape it using \.
An example for a static contact on an AOR:
contact=sip:172.16.1.1:5060;user=phone
Cheers,
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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
be seeing the a=imageattr in the SIP OK message?
It looks as though the passthrough for fmtp is indeed working but as
the imageattr attribute is currently unsupported/not used/not passed
through it is probably causing your resolution problem.
Cheers,
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Joshua Colp
Digium, Inc. | Senior Software
Matt Rabbitt wrote:
What would need to be changed in the source code to accommodate this?
Can the imageattr attribute be hard coded into
h264_format_attr_sdp_generate() in res_format_attr_h264.c?
A lot. Yes, you could hard code it.
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445
get a
backtrace[1] and file an issue[2] so we can take care of this? The
information you've provided in this email would also be useful. Thanks!
[1] https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace
[2] https://issues.asterisk.org/jira
--
Joshua Colp
Digium, Inc. | Senior Software
on purpose
3. Thought about the real world implications
4. Looked over required change
5. Committed to all branches
Cheers,
--
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
it? No, but I'm going to try. That's how much
I care. Can I do this for every issue? No. I work 8+ hours a day as it
is on Asterisk and also some on the weekends. So many issues, not enough
time. ^_^
Cheers,
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL
it?
Due to limitations within the Asterisk core you have to use the
PJSIP_DIAL_CONTACTS dialplan function[1].
[1]
https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+Function_PJSIP_DIAL_CONTACTS
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806
Yaron Nachum wrote:
Thanks Joshua,
I tried it already. That would generate a call to both AORs which is not
what I was looking for.
Isn't there a way to retrieve the AOR status from the dialplan?
Not currently.
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW
things
there's an AsteriskDev[2] account that I tweet things out on. Asterisk
releases, new additions, major changes, conferences Asterisk developers
are speaking at, that sort of thing.
[1] http://wiki.asterisk.org/
[2] http://twitter.com/AsteriskDev
Cheers,
--
Joshua Colp
Digium, Inc. | Senior
and further clarification/description! I was able
to reproduce and isolate the problem. I've fixed it in the 12 branch and
trunk.
Cheers,
--
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
appear to be using it correctly but that doesn't mean something isn't
up. Can you provide a SIP trace showing this for a further looksee?
Cheers,
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Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: www.digium.com
are
redirecting the channel which is executing ChannelRedirect (that
slightly made my head hurt). Switching should also make the variable
work as you desire.
Cheers,
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: www.digium.com
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Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: www.digium.com www.asterisk.org
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and they all get mapped back to a single entity (called an endpoint in
chan_pjsip).
I'm sorry this doesn't help you right now with chan_sip but I just
wanted to show that the future is bright and that we do listen. ^_^
Cheers,
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW
with the previous SIP stack so that I can
implement the API?
Depends on what you mean by cool new features. The ARI (Asterisk REST
interface) work is agnostic of channel drivers, it has no specialized
logic for any and any present channel driver can be used with it.
Cheers,
--
Joshua Colp
Digium, Inc. | Senior
this and still use the new AMI/API features as well as the
real time database config?
Yes.
Cheers,
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: www.digium.com www.asterisk.org
that can manage Asterisk in that way?
That's not an area I work in so I don't know what others have come up with.
I also understand that there is a real time database module.
You can store stuff in a database and have it used as a source for
configuration and such, yes.
--
Joshua Colp
Digium, Inc
config itself looks fine, what actually shows up on the console?
Cheers,
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: www.digium.com www.asterisk.org
originates from.
I'm afraid not, the only information in the dialplan even remotely
relating to SDP is the RTP address information.
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Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: www.digium.com www.asterisk.org
Is this a bug? Or something I can fix through config?
Hola,
Set transfer=no under the entries in iax.conf for the
peers/users/friends/etc in question, reload, retry, and see if that
changes the behavior. If it does then something involved may not like
IAX2 native transfers.
Cheers,
--
Joshua
creating a new connection
each time they run. How can I do this?
You can't, they are completely separate processes and code.
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Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: www.digium.com www.asterisk.org
CDR wrote:
I am trying to identify the module (*.so) that contains the Asterisk
Management Interface, so as to set noload=XXX.so in modules.conf. Any
idea?
There is no module, it's provided as core functionality. Disabling it
can be done in manager.conf
--
Joshua Colp
Digium, Inc. | Senior
) of the code exposing information in some
off nominal cases.
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: www.digium.com www.asterisk.org
.
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Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: www.digium.com www.asterisk.org
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.
While a channel may not be answered, it's still in existence.
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Joshua Colp
Digium, Inc. | Senior Software Developer
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Check us out at: www.digium.com www.asterisk.org
Notify-Subscribes outside
Asterisk scope ? How would you then qualify this experiment ?
I've got nothin' here.
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: www.digium.com www.asterisk.org
?
The realtime backends have no explicit knowledge of what they are being
used with, so any current backend can be used.
Cheers,
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: www.digium.com www.asterisk.org
modified chan_sip at all? I don't think it should be possible for it to
not put any media lines in...
Cheers,
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: www.digium.com www.asterisk.org
you have not loaded it either the endpoint IS
being used or you are actually using chan_sip.
Providing the configuration and console output would illuminate the
situation.
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Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out
=203.0.113.1
Done.
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Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: www.digium.com www.asterisk.org
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it?
The PJSIP module is not at feature parity with chan_sip, therefore that
functionality does not exist yet in it.
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Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: www.digium.com www.asterisk.org
://svn.digium.com/svn/asterisk/team/dlee/ASTERISK-22451-ari-subscribe/
Cheers,
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: www.digium.com www.asterisk.org
] https://issues.asterisk.org/jira
Cheers,
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: www.digium.com www.asterisk.org
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Nick Cameo wrote:
There is two way audio, it's just during ringing that this happens.
If you can put the SIP signaling and Asterisk console output up
somewhere then we can have a better idea of what Asterisk is being told,
and what it is doing.
--
Joshua Colp
Digium, Inc. | Senior Software
port (4569). As UDP is connectionless there are no
connections. What you see on the console is the *source* IP address and
port of the packets. It's possible that the Amazon stuff is sort of
NATting things to do connection tracking... but that's Amazon land, so
no clue really.
--
Joshua Colp
is still there using
allow=ulaw,alaw,g729.
As someone else mentioned the 'm' option is definitely in place.
Asterisk is doing exactly as it is told.
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: www.digium.com
Sean Darcy wrote:
On 09/10/2013 12:15 PM, Joshua Colp wrote:
Sean Darcy wrote:
Maybe a different question would be helpful. Let's assume no NAT; the
server is directly connected with an FQDN. Two iax devices register.
Does asterisk assign them different ports?
Asterisk does not assign ports
it.
As for your theory... not possible. Stuff can't react in that fashion
and change dial options mid-flow. Your output in a previous email also
showed the 'm' being specified to Dial, from an AGI. That's where you
should focus.
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan
... if you aren't being explicit with what username to
use for outgoing authentication then stuff like this can happen.
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Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: www.digium.com www.asterisk.org
:
username=home
And give it a go.
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: www.digium.com www.asterisk.org
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others do not run into the same issue.
Cheers,
PS: Thanks for giving Asterisk 12 a go and sorry you ran into this problem!
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: www.digium.com www.asterisk.org
://issues.asterisk.org/jira/browse/ASTERISK-22386
Despite there being no current progress on it keep checking back or add
yourself as a watcher.
Cheers,
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: www.digium.com
* is in JIRA. Even small stuff. As for the patch I
think that is personally how I would approach it, so if you could
provide it that would be appreciated.
Thank you!
-- Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out
and then
bridged the approach of using a Goto afterwards for ANSWER as well will
not work. You *must* use the h extension that was previously mentioned
to cover this case.
Cheers,
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us
,
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: www.digium.com www.asterisk.org
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(answered),NoOp(Call was answered)
exten = 534,102,NoOp(We reached step 102)
As I mentioned this won't work for ANSWER.
exten = h,1,NoOp(Dialing attempt got status ${DIALSTATUS})
will
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us
Larry Moore wrote:
On 31/07/2013 8:08 PM, Joshua Colp wrote:
Zoltán Fekete wrote:
Thank You Larry!
I have discussed with my provider. They are not able to insert the
T38MaxBitRate value into the sip answer. :(
https://gist.github.com/anonymous/6120148 (line 559)
That means we are not able
*. If a device requests SRTP
and it's not possible, the call will fail.
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: www.digium.com www.asterisk.org
ejabberd
The XMPP support is not tuned for Google Talk by any means, and the
voice part (chan_motif) supports the two Google derivatives and the
actual Jingle spec.
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out
in the situation. The asterisk.org Debian
packages have not been updated. The debian.org packages are done by
others, so I can't speak on that.
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: www.digium.com
.
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: www.digium.com www.asterisk.org
--
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sean darcy wrote:
I have a call on gv over motif. I try to bridge it to another call over
motif, but a different gv account, and I get congestion.
motif only handles one 1 channel at a time??
Motif itself has no imposed limitations, but that's not to say Google
Voice doesn't.
--
Joshua
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