r sending image from
> sip?
What do you mean by image?
--
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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excess file descriptor
usage in older versions, but it's so old that it's hard to remember.
Just to provide some scope of how many changes there are between even
1.8.11.0 and 1.8:
✔ jcolp@electron:~/development/asterisk/public [11|⚑ 1]> git diff
1.8.11.0..1.8 | wc -l
221688
That's
s, and imposes a 1024 open file
> limit count for some reason.
This is actually enforced by the system, not by Asterisk itself.
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Joshua Colp
Digium, Inc. | Senior Software Developer
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Check us out at:
ve to do a packet capture, look at the exchange in Wireshark, and
see how the negotiation flows. It requires a basic understanding of ICE.
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Joshua Colp
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ment.
Is there something nontrivial that needs to be done here other than just
recompiling/linking? If so, then I'm likely to run into it as well.
The way formats and codecs are defined within Asterisk was changed, as a
result code changes are required.
--
Joshua Colp
Digium, Inc. |
e ICE negotiation to see if it tried and
failed. After that would be looking at the DTLS negotiation. Asterisk
console output could provide some information.
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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.digiu
Richard Kenner wrote:
What is the proper version of the Siren7 codec to use for Asterisk 13.5.0?
Since there's nothing later, does the version for 12.0 work?
A Siren codec is not currently available and the one for 12 will not
work. I have no timeframe for when this might change.
--
J
early direct media. It is only once the call has
been established that it will be done.
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s
another INVITE, this time with port 1236
Asterisk sends the Trying to port 1236
You want to set "force_rport" to no.
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Joshua Colp
Digium, Inc. | Senior Software Developer
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Check us out at: www.digium.co
dtmf_mode = inband
device_state_busy_at = 32
chan_pjsip has few defaults. As a result the above endpoint has no
codecs configured. Therefore the SDP negotiation fails. Adding
"allow=ulaw" should allow it to work.
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davi
s" to your endpoint. This
will cause it to reuse the existing connection established from the
phone. Generally the port provided by the phone is not reachable.
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Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at:
ill Asterisk send just one message SIP 183 to the caller, with
some kind of generic SDP message?
Asterisk isn't a proxy, so it won't forward all 3 and it won't forward
media from all 3. Right now the Dial application is simple and just
doesn't forward media in this scenario.
acket with a zero datalen anyway?
This is an interpolated frame from func_jitterbuffer. It's part of
packet loss concealment. What scenario exposed this?
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check
https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+ManagerAction_DBGet
[2] https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Asterisk+REST+API
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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.ast
.
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
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can be provided as part of the endpoint id:
[t...@theworld.com]
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
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___
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
Ludovic Gasc wrote:
2015-05-21 17:59 GMT+02:00 Jean-Denis Girard mailto:jd.gir...@sysnux.pf>>:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Le 21/05/2015 00:16, Joshua Colp a écrit :
> If CCSS is needed then the only option is to use chan_sip. The
> chan_
Jean-Denis Girard wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Le 21/05/2015 00:16, Joshua Colp a écrit :
If CCSS is needed then the only option is to use chan_sip. The
chan_pjsip module does not implement CCSS in any way.
Is CCSS support planned for PJSIP? chan_sip is in "ext
does not implement CCSS in any way.
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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Jean-Denis Girard wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Le 20/05/2015 00:50, Joshua Colp a écrit :
It looks like this is an incoming leg, in which case that information
isn't available. There is no association of an AOR and Contact on
incoming legs (it MAY be possible to d
, in which case that information
isn't available. There is no association of an AOR and Contact on
incoming legs (it MAY be possible to deduce but it certainly wouldn't
work in all cases). Since you specify one explicitly on outgoing, that's
when it is available.
--
Joshua C
Andrew Martin wrote:
- Original Message -
From: "Joshua Colp"
To: "Asterisk Users Mailing List - Non-Commercial
Discussion"
Sent: Wednesday, May 13, 2015 10:10:25 AM
Subject: Re: [asterisk-users] "Retransmission Timeout" results in dropped calls
a
opped?
The traffic is between the phone and Asterisk. As to why, I have no
idea. The packets aren't getting to Asterisk - that's all I can say. I
doubt it's network turbulence. Likely getting lost/blocked somewhere.
--
Joshua Colp
Digium, Inc. | Senior Software Developer
44
elation to directmedia between your
version and latest 11.
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Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
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Andrew Martin wrote:
- Original Message -
From: "Joshua Colp"
To: "Asterisk Users Mailing List - Non-Commercial
Discussion"
Sent: Monday, May 11, 2015 12:32:06 PM
Subject: Re: [asterisk-users] "Retransmission Timeout" results in dropped calls
a
up and terminates the dialog.
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Digium, Inc. | Senior Software Developer
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symack wrote:
Hello Everyone,
We have most of the modules commented out. Can someone please let me
know which modules needed to be included for Playtones?
The PlayTones application is in the app_playtones module.
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW
ql menuselect.makeopts
After looking at the issue this appears to be a duplicate of an existing
one[1] and a known issue.
[1] https://issues.asterisk.org/jira/browse/ASTERISK-21228
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
C
/browse/ASTERISK-24910
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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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output on
the CLI? As well - you have one of the parameters incorrect above. It's
timers_sess_expires, not timers_sess_expiries. If that is incorrect in
your configuration this would be considered invalid and thus it would
not load.
Cheers,
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Joshua Colp
Digium, Inc. | Senior Sof
d be a bug. File an issue on the issue tracker[1].
[1] https://issues.asterisk.org/jira
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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 &a
https://bugzilla.mozilla.org/show_bug.cgi?id=1147919
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Digium, Inc. | Senior Software Developer
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Check us out at: www.digium.com & www.asterisk.org
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name it the same or not is up to the SIP client
registering. Some allow you to specify the AOR you are registering
against. Some assume that the username you are authenticating as is the
same as your AOR.
Cheers,
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Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW
opened the firewall so RTP can be received at the Asterisk that
advertises the public IP?
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 & ww
s like you need the EVAL dialplan function[1].
https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Function_EVAL
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 & ww
on the console when chan_sip is loaded?
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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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is no explicit configuration required except ensuring hints are
available for the extensions you wish to subscribe to. The subscribing
device does need to support the standard as well.
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Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Sonny Rajagopalan wrote:
[sonnyGW1]
type=identity
endpoint=sonnyGW1
match=65.254.44.194
You want type=identify, not type=identity.
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
o out. You can also remove the explicit transport from
the endpoint.
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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.ast
sed on applications built upon Asterisk.
Cheers,
[1] https://wiki.asterisk.org/wiki/pages/viewpage.action?pageId=29395573
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digi
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?
--
quot; from your endpoints.
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Joshua Colp
Digium, Inc. | Senior Software Developer
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f you use TCP then set rewrite_contact=yes so it'll reuse the
established TCP connection.
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Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & ww
Nick Awesome wrote:
Hay guys, got trouble with registration with cisco 7975
The "force_rport" option is incompatible with Cisco, it needs to be
explicitly set to no in the endpoint.
Cheers,
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Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Hunt
ne phone rings, seems like it take
first one as Dial app by default, is there way to fix this?
There is no way to directly do this. The best option is to use a Local
channel into the dialplan which dials instead. Once answered everything
should fall into place.
--
Joshua Colp
Digium, Inc. |
sterisk/branches/?view=log&pathrev=375893
[2] https://issues.asterisk.org/jira/browse/ASTERISK-20032
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check
for filing an issue are here[1] and include links for
submitting a patch. I know of noone currently working on it.
[1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines
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Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806
. I know
you said that it *should* work, with no guarantee, which I'm fine with. I
just want to make sure I don't have a possible misconfiguration issue.
Your configuration looks correct at first glance. Without labbing it up
myself I don't really have anything else to suggest.
fix it would be to add an OutgoingSpoolFailed extension
for each priority so that Swift isn't invoked. You could also use GotoIf
in the first priority and go elsewhere if the extension is
OutgoingSpoolFailed. There exist many options.
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Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan D
others are not. I was curious
what the status of those objects are.
It "should" be possible with recent changes that have been done. I
personally have not tried it, and it would still require a reload to
pick up changes regardless.
Cheers,
--
Joshua Colp
Digium, Inc. | Senior Software
l channel and has no formats, that will fail. Since your dialplan
logic also has it go in a loop it just goes 'round and 'round.
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
Sunny wrote:
Hi list,
Kia ora,
How do I make Asterisk 13 (using PJSIP channel) to handle PUBLISH sent
from the phones?
Noone has implemented this functionality. The only way to have it handle
it would be to write an implementation.
Cheers,
--
Joshua Colp
Digium, Inc. | Senior Software
f folks on OSX doing the odd thing.
Cheers,
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Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
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Sunny wrote:
Yeah, that's the case. The endpoint is listed in pjsip show endpoints.
Glad to hear that! I've created an issue[1] to track the underlying
problem there.
Cheers,
[1] https://issues.asterisk.org/jira/browse/ASTERISK-24748
--
Joshua Colp
Digium, Inc. | Senio
dpoints before it can...
leading to an error message that eventually gets resolved.
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queried first followed by the configuration file. If
you want to change the order then swap their placement.
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
Matt Hoskins wrote:
Hello,
Kia ora,
I’m currently evaluating asterisk 13 (Currently on 11). We’re testing
the migration from SIP to PJSIP. Is there a way to alias the SIP
channeltype to PJSIP when exlusively using pjsip?
There is not.
Cheers,
--
Joshua Colp
Digium, Inc. | Senior Software
Matt Hoskins wrote:
Joshua,
Regarding Outbound Registrations in realtime and a reload. Does it require
a "pjsip reload" or full asterisk restart?
It requires a module reload res_pjsip_outbound_registration.so
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davi
an lead to errant behavior."
Which objects and is this still true in 1.13.1 ?
Yes, still true for transports. Outbound registrations, though, have
been tweaked so that they can exist in realtime but require a reload to
be picked up. Those are the two that come to mind.
Cheers,
--
Jos
uth to use is pulled
from there.
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Digium, Inc. | Senior Software Developer
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what you mean.
But I bet you're now going to say, those small amounts are going to add up..
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us o
d
the length of ps_endpoints.auths = 40. This suggests there are scenarios
where there
are aors, without corresponding auth. Can you mix dynamic and static
AORs within one
endpoint, and would there be a use case for that?
You can mix however you want.
--
Joshua Colp
Digium, Inc. | Senior Software Deve
does aor have a max_contacts value?
And where do phone registrations fit in, where are those kept anyway?
Cheers,
[1] https://wiki.asterisk.org/wiki/display/AST/Configuring+res_pjsip
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
problems. I've rarely (if ever) come across an ALG
(that's what that is) that didn't break something.
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
. What is the
full SIP signaling?
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Digium, Inc. | Senior Software Developer
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use a default port of 5060.
I also believe I've covered your origination issue in a separate email.
Your dial string should be:
PJSIP/800...@outbound.vitelity.net
Cheers,
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Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check u
tried that. However, that did not work either.
Authentication controls authentication, it doesn't control how PJSIP
associates traffic with a specific endpoint. They are separate things.
I think before we get into config we need to see the dial string for
your origination.
Cheers,
pected state and it asserts. It's safe to
do as you've done since the operation will do nothing, but that's not a
viable fix to push upstream as it's a result of an implementation detail
with chan_pjsip. The fix will occur in chan_pjsip once the issue is handled.
--
Joshua Colp
D
ing that to get
profile information so it can provision itself and become aware if its
configuration changes.
What softphone is it? It sounds like rather odd behaviour.
It's odd to me that this is a required RFC. Is this softphone written to
be used against a specific platform?
Cheers,
-
correct, there is no support for ua-profile and a 489 is the
correct response. This shouldn't cause any problems though, unless the
softphone absolutely requires that RFC to be implemented in which case
it won't work. The only workaround for likely be to implement it if that
is the c
.
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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a dialplan function for
querying for AOR information.
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
--
__
Matt Hoskins wrote:
Of course, I left out a detail that may (or may not change) the answer.
I'm using the external chan-sccp-b sccp module, not the chan_skinny
bundled with asterisk.
Still doesn't matter. Provided it works with res_xmpp it'll work with
the new SIP method.
Che
Matt Hoskins wrote:
Before I go down a rabbit hole, does the mwi publish/subscription work for
non SIP phones?
Yes. SIP is simply used as the transport mechanism. It works pretty much
the same as res_xmpp except without needing an XMPP server.
Cheers,
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Joshua Colp
Digium, Inc. | Senior
non-fatal situation.
Previously it would reject it, since we don't support key lifetimes.
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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.ast
time unsupported: %s\n", attr);
is remove the:
continue;
Afterwards.
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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 & ww
arned.
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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James Lamanna wrote:
Hi Matt,
So this actually works (haven't had a chance to try it)?
SET VARIABLE CHANNEL(musicclass) default
Because musicclass is piece of channel information.
Referencing ${musicclass} is not the same thing.
It should indeed work, yes.
--
Joshua Colp
Digium
king me lose my hair).
I would suggest using the latest version of 11 (as older versions will
not work with current browsers). As well do you have the uuid
development library installed? If not pjproject won't be built and you
won't have ICE support which will yield exactly this result
Sets, 502, 5) exited non-zero on 'DAHDI/4-1'
This means that whatever was executing either encountered an error or
the channel was hung up, which is when dialplan execution stops. This is
normal as I previously mentioned.
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Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Da
quent dialplan
logic does not execute. You need to place the rest in the 'h' extension
which is executed upon hangup.
Cheers,
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Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Ch
de chooses the wrong one using
automatic logic.
What do you mean by "assign multiple IPs to an endpoint".
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in
the code. Since usage of the astdb is up to everything else there is
nothing stored in it until other stuff puts something there.
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
SIP/9001-",
"PJSIP/9002,20") in new stack
-- Called PJSIP/9002
== Everyone is busy/congested at this time (1:0/0/1)
-- Auto fallthrough, channel 'PJSIP/9001-' status is
'CHANUNAVAIL'
What is shown if you do "pjsip set logger on&qu
at... works...
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Joshua Colp
Digium, Inc. | Senior Software Developer
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me, coming back to chat_sip :|
How will you do this in chan_sip? The behavior between the two is the
same, despite the configuration being different.
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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.
Nick Awesome wrote:
Thats because I call from one to other
Then no, you can only match based on IP address. This also applies to
chan_sip. You have to send both to the same context and then within
there you can differentiate them based on the dialed number.
--
Joshua Colp
Digium, Inc
0*50*]
type=endpoint
context=did-2
disallow=all
allow=ulaw
If this is not correct then you can only match once based on the source
IP address currently.
Cheers,
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Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us ou
as to have license), but not
sure if community versions offer video calls at all.
Video transcoding is both usually patent encumbered as well as
computationally expensive. Asterisk supports passing through the video
untouched, but that's about it.
Cheers,
--
Joshua Colp
Digium, Inc
xt between dashes.
The configuration parser can do a lot of things. Out of curiosity
amongst those reading this - how many of you know about templates?
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & ww
ow more about, Mitch?
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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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t. I personally use
the SVN version of pjproject to ensure that Asterisk can compile against
it at all times. Provided you use the 2.x series then it's fine.
[1] http://www.pjsip.org/
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806
that
and include it in my email. Right now they just all bounce CHANUNAVAIL
which is expected.
If you are using Asterisk 11 or above you can use the hangup causes
functionality[1] to get more detail including the protocol specific reason.
Cheers,
[1] https://wiki.asterisk.org/wiki/display/AST/Hang
?
There is a "transmit_silence" option in asterisk.conf which will cause
Asterisk to transmit silence during certain actions (such as recording).
If you set this to yes it should resolve your issue.
Cheers,
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW -
/AST/Asterisk+12+ARI
[3] http://lists.digium.com/cgi-bin/mailman/listinfo/asterisk-app-dev
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.co
D'Arcy J.M. Cain wrote:
On Sun, 10 Aug 2014 09:56:48 -0300
Joshua Colp wrote:
Of course. Thanks. The odd thing is that that comes from the
distribution. Any idea where that template is meant to be used?
It's not as far as I know. It's just being used as a method so the
val
D'Arcy J.M. Cain wrote:
On Sun, 10 Aug 2014 07:57:31 -0300
Joshua Colp wrote:
[directories](!)
astetcdir => /usr/local/etc/asterisk
etc...
Remove (!) from the name. That marks it as a template. Unless you
remove it it won't be used.
Of course. Thanks. The odd thing is th
local/etc/asterisk
etc...
Remove (!) from the name. That marks it as a template. Unless you remove
it it won't be used.
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium
ey use that to determine who you are trying to
authenticate as. If you require this to be set then caller id
information has to be transported in a different manner (RPID or PAI).
Cheers,
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
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