ial cause of failure?
--
Joshua C. Colp
Digium - A Sangoma Company | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org
--
_
-- Bandwidth and Col
for chan_sip is provided by the community and
there is no time frame on when (or if) such a thing would be looked into so
keep that in mind.
--
Joshua C. Colp
Digium - A Sangoma Company | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.
I connection to same app name then above problem solved and
> after some call i found same above issue,
> and this issue rise when call load is high, so i don't understand what
> is problem, so please give solution of this problem.
Have you looked at the actual ARI events that are
ike that in Asterisk. It would be
trying to do server side three way calling, which is not supported like that.
--
Joshua C. Colp
Digium - A Sangoma Company | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 -
g is done. This
can be done using the 'g' option to Dial[1] which continues dialplan
application after the outgoing call leg hangs up. You could even send
information as SIP headers if need be so S sees the info.
[1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+16+Application_Dia
Asterisk version from what I can see, and stream
behavior between 13 and 16 differs (as 16 understands streams) which could
contribute to the behavior.
--
Joshua C. Colp
Digium - A Sangoma Company | Senior Software Developer
445 Jan Davis Drive NW -
tuff just informs it of what is going on and it then takes care of the
rest. ARI should behave the same way in regards to generation of them for
dialing and other things.
--
Joshua C. Colp
Digium - A Sangoma Company | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - U
On Thu, May 30, 2019, at 11:30 AM, Jonas Kellens wrote:
> Hello
>
> is this mailing list still active ?
Seems like it. :D I responded previously. Many people have moved to
Discourse[1] though and it sees more activity.
[1] https://community.asterisk.org/
--
Joshua C. Colp
Digium -
nnel.
Any such functionality would be documented on the wiki[1].
[1]
https://wiki.asterisk.org/wiki/display/AST/Asterisk+16+Configuration_res_pjsip
--
Joshua C. Colp
Digium - A Sangoma Company | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at:
bug in a case. The chan_sip module is community
supported so it does not see a lot of change.
The chan_pjsip module is maintained and in regards to video is something that
the team at Sangoma who work on Asterisk daily use for video meetings.
--
Joshua C. Colp
Digium - A Sangoma Company
what we would specify in that scenario automatically.
[1]
https://wiki.asterisk.org/wiki/display/AST/Asterisk+16+Function_PJSIP_SEND_SESSION_REFRESH
--
Joshua C. Colp
Digium - A Sangoma Company | Senior Software Developer
445 Jan Davis Drive NW - Huntsville
it for the voicemail app to cause the event to get emitted.
>
>
> Is this possible? AMI or asterisk command?
Do you mean something like the MailboxCount AMI action[1]?
[1]
https://wiki.asterisk.org/wiki/display/AST/Asterisk+16+ManagerAction_MailboxCount
--
Joshua C. Colp
Digium - A Sango
On Thu, May 16, 2019, at 2:57 PM, Joshua C. Colp wrote:
> On Thu, May 16, 2019, at 2:54 PM, Nick Olsen wrote:
> > Hello all,
> >
> > I'm migrating a box from PJSIP with normal Flatfiles to ODBC/Realtime,
> > Also 16.0.1 to 16.3.0. After adding a few peers to
Any other tweaks I can make to asterisk to speed this up (Not really
> looking to match based on other objects, like header or username).
[1] https://wiki.asterisk.org/wiki/display/AST/Sorcery+Caching
--
Joshua C. Colp
Digium - A Sangoma Company | Senior
ing built into Asterisk itself for it. The sqlite
library calls are directly used and their results provided.
--
Joshua C. Colp
Digium - A Sangoma Company | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www
support a CIDR or wildcard or multi-ip format for the
> host= line in sip.conf?
The chan_sip module does not support this. The chan_pjsip module supports this
in the identify section.
--
Joshua C. Colp
Digium - A Sangoma Company | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, A
ble would not exist. You'd
need to calculate it yourself.
--
Joshua C. Colp
Digium - A Sangoma Company | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org
--
_
o differentiate. Some may use a similar mechanism to the
line option. Some run multiple SIP transports on different ports for each
account so they can differentiate based on where it came in on. Some look at
the request URI coming i
gards to different outbound registrations. It
requires the remote server to adhere to the SIP RFC and report back some data
we give in our Contact, so you have to test it and see if it works.
[1]
https://blogs.asterisk.org/2016/01/27/the-pjsip-outbound-registration-line-option/
--
Joshua C. Co
On Wed, Apr 17, 2019, at 2:06 PM, Brian J. Murrell wrote:
> On Wed, 2019-04-17 at 11:56 -0400, Joshua C. Colp wrote:
> > On Wed, Apr 17, 2019, at 12:51 PM, Brian J. Murrell wrote:
> > >
> > > I can add it onto the end of the variable in the Dial() command:
> > &g
On Wed, Apr 17, 2019, at 12:51 PM, Brian J. Murrell wrote:
> On Wed, 2019-04-17 at 10:04 -0400, Joshua C. Colp wrote:
> >
> > You specify the transport in the SIP URI. For example:
> >
> > sip:t...@example.com;transport=tcp
>
> Hrm. This is probably going to b
specify the transport in the SIP URI. For example:
sip:t...@example.com;transport=tcp
This will limit to TCP, and depending on the resolution of "example.com" use
IPv6 or IPv4. The ICE candidates should then be both IPv6 and IPv4 since a
transport is not explicitly specified.
--
Jos
t;
> I'm expecting gv-voice to be the "matching endpoint". The INVITE has
> gv-voice as the "Contact:" . Isn't this the "Username" in pjsip "auth" ?
Nope. The Contact is never considered for that. The From username is what is
matched for an endpoin
ou using ICE? If not I'd suggest setting
"icesupport" to "no" in rtp.conf and seeing if that helps. It seems to be
trying to get ICE candidates which is taking a period of time on your system.
It could be an unresponsive STUN server, or interfaces which go nowhere bloc
ally say anything. The
SIP traffic, "pjsip set logger on", and normal verbose information mixed in
would provide more.
As well, does /etc/hosts contain an entry for the system itself using its own
hostname?
--
Joshua C. Colp
Digium - A Sangoma Company | Senior Software Developer
445 Jan
I'd suggest
providing console output somewhere so we can see precisely what is going on.
--
Joshua C. Colp
Digium - A Sangoma Company | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check u
email though if you
bump up the debug it'll tell you precisely what it's doing (resolving hostname
blah for an A record, etc).
--
Joshua C. Colp
Digium - A Sangoma Company | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com &a
core set debug 9) and the resolver will tell you what it is
trying to do.
--
Joshua C. Colp
Digium - A Sangoma Company | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www
from moving from chan_sip to chan_pjsip.
What version of Asterisk? That will change the answer as 13 uses the built-in
PJSIP DNS resolver, while 16 uses our own implementation.
--
Joshua C. Colp
Digium - A Sangoma Company | Senior Software Developer
445 Jan Davis Drive NW - Huntsville,
he
call is from a Queue. You could even have a dialing context per queue to know
which one if you wanted.
--
Joshua C. Colp
Digium - A Sangoma Company | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, A
On Tue, Apr 2, 2019, at 9:06 PM, Sungtae Kim wrote:
>
> On 4/3/19 1:29 AM, Joshua C. Colp wrote:
> > On Tue, Apr 2, 2019, at 8:26 PM, Matthew Jordan wrote:
> >>
> >> On Tue, Apr 2, 2019 at 4:18 PM Joshua C. Colp wrote:
> >>> On Tue, Apr 2, 2019, at 8:15
On Tue, Apr 2, 2019, at 8:26 PM, Matthew Jordan wrote:
>
>
> On Tue, Apr 2, 2019 at 4:18 PM Joshua C. Colp wrote:
> > On Tue, Apr 2, 2019, at 8:15 PM, BJ Weschke wrote:
> > > I get the desired use case to run app_amd from within a Stasis
> > > application,
ch as Queue, Dial, ConfBridge,
Playback, Record or some others really make sense.
--
Joshua C. Colp
Digium - A Sangoma Company | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org
___
n why app_queue would be executed from ARI? What value does ARI
bring in that regard?
--
Joshua C. Colp
Digium - A Sangoma Company | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org
__
get them installed please?
PJSIP does not use those dialplan applications. They are for the chan_sip
module. PJSIP instead uses the PJSIP_HEADER[1] dialplan function.
[1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+16+Function_PJSIP_HEADER
--
Joshua C. Colp
Digium - A Sangoma Company |
quot;norefersub" Supported
occurs before it happens. You could only limit it globally.
[1] https://www.ietf.org/rfc/rfc4488.txt
--
Joshua C. Colp
Digium - A Sangoma Company | Senior Software Developer
445 Jan Davis Drive
t; packet Asterisk sends to Cisco? Perhaps Cisco sees the norefersub, but
> not the Refer-Sub header. It may interpret the lack of a Refer-Sub
> header in the REFER incorrectly?
It's entirely possible it might. It doesn't hurt to try.
--
Joshua C. Colp
Digium - A S
e REFER
itself. I don't believe we set that on the REFER we produce, we only care if we
receive a REFER with it in place.
--
Joshua C. Colp
Digium - A Sangoma Company | Senior Software Developer
445 Jan Dav
ctivity on DAHDI channels, or parking lots).
>
> What other configuration is required to get Qualify working? Thank you
> in advance.
The OPTIONS and the SUBSCRIBE are unrelated. If an OPTIONS goes out and gets a
response, then it's working.
The SUBSCRIBE is the phone attempting
rhaps asterisk developers would consider eventually add the
> rewrite support in asterisk?
It's certainly something that could be done, I just don't know of anyone
working on it right now or if anyone plans to.
--
Joshua C. Colp
Digium - A Sangoma Company | Senior Software Devel
et with
some pushback so this is another avenue.
--
Joshua C. Colp
Digium - A Sangoma Company | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org
--
_
--
e
pjsip show CLI commands for each thing)
--
Joshua C. Colp
Digium - A Sangoma Company | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org
--
_
)
The most active deployment, or a few deployments, is fine.
--
Joshua C. Colp
Digium - A Sangoma Company | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org
--
__
t be a template?
The (!) has to be removed from the end of [directories] or else it won't take
effect.
--
Joshua C. Colp
Digium - A Sangoma Company | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 -
On Fri, Mar 1, 2019, at 5:09 PM, Brian J. Murrell wrote:
> On Fri, 2019-03-01 at 15:54 -0500, Joshua C. Colp wrote:
> >
> > That's correct. You'd either need to retrieve the line parameter from
> > the outbound registration or forge the source IP address,
>
&g
On Fri, Mar 1, 2019, at 4:51 PM, Brian J. Murrell wrote:
> On Fri, 2019-03-01 at 15:41 -0500, Joshua C. Colp wrote:
> >
> > I don't understand what you mean. Your ITSP has stated that they
> > don't want you to do authentication with them, so you can
On Fri, Mar 1, 2019, at 4:33 PM, Brian J. Murrell wrote:
> On Fri, 2019-03-01 at 14:15 -0500, Joshua C. Colp wrote:
> > you can try line functionality on the outbound registration which
> > may or may not work[2] (requires the upstream to adhere to the RFC,
> > which not all
On Fri, Mar 1, 2019, at 3:56 PM, Brian J. Murrell wrote:
> On Fri, 2019-03-01 at 14:15 -0500, Joshua C. Colp wrote:
>
> [itsp-endpoint](!)
> type=endpoint
> transport=transport-udp
> context=from-itsp
> message_context=messages
> disallow=all
> allow=ulaw
> from
2016/01/27/the-pjsip-outbound-registration-line-option/
--
Joshua C. Colp
Digium - A Sangoma Company | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org
--
__
in
> rtp.conf, or did I just get lucky that by chance the clients happened to
> use these ports when I made the test calls?
It's exchanged as part of call setup using SDP. SDP specifies where media
should be sent, the codecs that can be used, and also controls hold/unhold.
Each side
of media, except for
circumstances where you know for sure the source.
Note that RTP ports in Asterisk aren't open all the time and only listen when a
call is using it, and they also learn the source of media - blocking out other
sources.
--
Joshua C. Colp
Digium - A Sangoma Company
ports. It does not, and can not, control the remote endpoint. It's up to the
endpoint and if NAT Is involved the router as to what source port is used for
media originating from the endpoint.
--
Joshua C. Colp
Digium - A Sangoma Company | Senior Software Developer
445 Jan Davis D
On Sat, Feb 23, 2019, at 8:06 AM, hw wrote:
> On 2/22/19 7:56 PM, Joshua C. Colp wrote:
> > On Fri, Feb 22, 2019, at 2:48 PM, hw wrote:
> >>
> >> Hi,
> >>
> >> when trying to use SRTP, I can see UDP traffic from phones to the
> >> asterisk s
ia with SRTP is not supported. All media when SRTP goes through
Asterisk.
--
Joshua C. Colp
Digium - A Sangoma Company | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org
--
__
like chan_sip did. There are cases where it can
still block though.
--
Joshua C. Colp
Digium - A Sangoma Company | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digiu
On Fri, Jan 18, 2019, at 1:06 PM, Olivier wrote:
>
>
> Le ven. 18 janv. 2019 à 17:30, Joshua C. Colp a écrit :
> >
> >
> > >
> > > You mean with a softphone you can't select a single (or several) video
> > > among those availab
not really a name or anything, except for our content type that we came
up with. It's not defined as a standard or an RFC. Within ConfBridge the
messages are exchanged using JSON between Asterisk and client.
--
Joshua C. Colp
Digium - A Sangoma Company | Senior Software Developer
445 Ja
rt the JSON payload that Asterisk is using in ConfBridge. This
conveys additional information that a straight up chat message wouldn't.
--
Joshua C. Colp
Digium - A Sangoma Company | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium
help.
There's a blog post which shows how it is supposed to work[1]. It expects the
channel to be created, then both put into the bridge, and then dialed. This
also requires Asterisk 14 or above to operate. What version are you using?
[1] https://blogs.asterisk.org/2016/08/24/asterisk-14-ari-c
On Tue, Jan 15, 2019, at 1:17 PM, Brian J. Murrell wrote:
> On Tue, 2019-01-15 at 12:01 -0500, Joshua C. Colp wrote:
> >
> > The chan_sip module has this implemented under the "nat" option using
> > "comedia" as I recall.
>
> Yeah. The help for wh
_sip or chan_pjsip to know/care, as it's
just SIP. You'd need to look at the SIP signaling and the SDP to understand
what is happening.
--
Joshua C. Colp
Digium - A Sangoma Company | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digiu
Broadsoft has done from their platform.
--
Joshua C. Colp
Digium - A Sangoma Company | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org
--
_
-- B
(between initiation and answer)
> is recorded in a separate CDR?
An excellent question. Unlike in the past versions calls can actually generate
multiple CDRs because CDRs now represent the flow of communication between
things.
Providing the actual CDR records that were generated as well a
console.log(err);
> });
>
I believe you are accessing the snapshot, essentially, of the bridge at the
time it was created in which case there would be no channels. You would need to
retrieve an up to date snapshot to get the current
ith my cellphone ringing for at least 10 seconds
> and ringing heard on the Yealink connected to the asterisk - e. g.
> completely wrong:
This is the way it is supposed to work[1], but it ultimately depends on your
dialplan. Are you using Local channels? Are you doing Answer in the dialplan?
Wha
sed to install it.
--
Joshua C. Colp
Digium - A Sangoma Company | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org
--
_
-- Bandwidth and Colocatio
On Thu, Dec 27, 2018, at 1:07 PM, Michael Maier wrote:
> Hi!
>
> I just want to say, that 13.24.1 doesn't fix the problem described in
> the posts above.
You're going to need to file an issue[1] with traces and actual configuration.
[1] https://issues.asterisk.org/ji
y comes to mind that would be accomplished by sending a 183
without SDP but there may be cases on the internet.
--
Joshua C. Colp
Digium - A Sangoma Company | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com &am
. I'd
therefore suggest raising an issue[1] with the SIP trace.
[1] https://issues.asterisk.org/jira
--
Joshua C. Colp
Digium - A Sangoma Company | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org
--
lt of menuselect is stored in the "menuselect.makeopts" file. Copying
that over should save some time, and if menuselect needs to be controlled in a
scripted fashion it can also be done so[1].
[1]
https://wiki.asterisk.org/wiki/display/AST/Using+Menuselect+to+Select+Asterisk+Options
manipulated using the normal core mechanisms[1].
[1]
https://wiki.asterisk.org/wiki/display/AST/Manipulating+Party+ID+Information#ManipulatingPartyIDInformation-REDIRECTINGdialplanfunction
--
Joshua C. Colp
Digium - A Sangoma Company | Senior Software Developer
445 Jan Davis Drive NW - Hun
On Fri, Dec 7, 2018, at 9:54 AM, hw wrote:
> On 12/07/2018 02:25 PM, Joshua C. Colp wrote:
> > On Fri, Dec 7, 2018, at 9:19 AM, hw wrote:
> >>
> >> Hi,
> >>
> >> is cdr logging using odbc buggy? I'im only getting an error
> >> "c
nf works just
> fine, and using the database for sippeers also works.
This message is output when the "dsn" value provided in cdr_odbc.conf does not
match a dsn/class (context name) configured in res_odbc.conf
You should confirm they match and if still encountering a problem the
On Thu, Dec 6, 2018, at 1:37 PM, Sebastian Damm wrote:
> Hi,
>
> On Thu, Dec 6, 2018 at 2:44 PM Joshua C. Colp wrote:
> > Nope. No specific reason. The ones there are what most people would use,
> > and what those not as familiar with telephony would understand.
>
I could submit a patch for more hangup reasons, but before doing so,
> is there a reason that there are only those five hangup reasons?
Nope. No specific reason. The ones there are what most people would use, and
what those not as familiar with telephony would understand.
--
Joshua C. Colp
Dig
On Tue, Nov 27, 2018, at 8:13 AM, Administrator TOOTAI wrote:
> Le 27/11/2018 à 12:13, Joshua C. Colp a écrit :
> > On Tue, Nov 27, 2018, at 5:49 AM, Administrator TOOTAI wrote:[...]
> >>
> >> [TOOTAiAudio]
> >> ;
> >> ; Call our gateway
> >&g
port type like was done in chan_sip. You can set an explicit one to use,
but in some cases this may not get used (SIP responses won't use it).
--
Joshua C. Colp
Digium - A Sangoma Company | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
C
] the PJSIP_HEADER dialplan function has to be
executed on the PJSIP channel itself, not the calling channel. You need to use
a pre-dial handler and invoke it there.
[1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Function_PJSIP_HEADER
--
Joshua C. Colp
Digium
to find the correct action for sending events. Should this
> action listed in "actions:commands"? How can i sent the
> "QueueMemberAdded" Event? and what is the correct format?
AMI does not allow you to send events like this. They can only be generated and
sent by the C c
On Fri, Oct 12, 2018, at 7:22 AM, Dmitry Melekhov wrote:
> 12.10.2018 14:10, Joshua C. Colp пишет:
> > On Fri, Oct 12, 2018, at 3:35 AM, Dmitry Melekhov wrote:
> >> Hello!
> >>
> >> Just upgraded asterisk from 13 to 16 and found that php-agi library is
> &g
to execute (no such command, invalid syntax etc.) now
return an Error response instead of Success.
--
Joshua C. Colp
Digium - A Sangoma Company | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
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