did you install it - from a
package or pip? A change may have gone in which somehow requires a newer
version.
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they can't be
downloaded any longer, however, is a bug. We'll get this fixed up as
soon as possible.
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
provided your feedback to Rusty Newton, who is spearheading
documentation updates, and he'll work on including them in the wiki.
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
Marek Červenka wrote:
and what about
https://www.asterisk-blog.com/2016/02/17/odbc_gutting/
While not in the email these are listed in the CHANGES and UPGRADE.txt
file. Going forward we'll try to ensure we include such things in the
release notes as well.
--
Joshua Colp
Digium, Inc
support matching based on the IP
address+port that a device has registered from. You can only explicitly
configure this right now.
[1] https://gerrit.asterisk.org/#/c/2373/
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Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us
that registers after
receiving push notification) to a progressing (not answered) call
The Asterisk dialing process itself does not allow this. Once channels
are dialed you can't add. You'd need to send the push notification, wait
a period of time, and then do the Dial.
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Joshua Colp
Digium, Inc
xt_ext},${next_step})
Will this work? Does Asterisk evaluate expressions like this, or does it
expect literals?
It most certainly will work. It evaluates on use.
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Digium, Inc. | Senior Software Developer
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multiple registrations
to a single AOR and with some changes to dialplan all are dialed when
called.
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u have some sort of helper
which is opening up the right firewall ports and when TLS is in use it
can't see the traffic and thus doesn't.
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445 Jan Davis Drive NW - Huntsville, AL 35806 - US
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there actually is a
firewall.
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, if it's the
external then we've told the phone to send it to the right place. After
that do a packet capture and see if the packets are arriving on the
machine. If not then look outside the machine at things.
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Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW
Chirag Desai wrote:
Joshua Colp wrote:
Have you done a packet capture to see if the RTP from the remote device
is hitting the machine to narrow things down?
Nope. When I run with RTP encryption on it seems that rewrite_contact
does not work in PJSIP.
When I turn
on the other side? What is
the full SIP trace?
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environment so the
endpoint and the asterisk server *could* reach each other by the
private IP ,but I am actually trying to avoid this with a proper
configuration since my real users will not be on any VPN, mostly.
What version of 13 are you also using?
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Digium, Inc. | Senior Software
any then check to see if
they are being blocked by a firewall, and that the SDP sent to the
device contains the public IP address.
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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.aster
on is used to control this for RTP. When set to
yes media will be sent to the source IP address+port of the received RTP.
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
ourage those to be contributed[1].
Cheers,
[1] https://wiki.asterisk.org/wiki/display/AST/Patch+Contribution+Process
--
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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 &
://issues.asterisk.org/jira
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445 Jan Davis Drive NW - Huntsville, AL 35806 - US
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an
issue[1] so we can provide more information for that warning or
potentially squash it for CNG.
[1] https://issues.asterisk.org/jira
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to the outgoing side, which may mean that the length of digits
or something else produced by the softphone is not liked by it.
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
/wiki/display/AST/Patch+Contribution+Process
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; <--- only this line was changed.
No, nothing else is needed.
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Check us out at: www.digium.com & www.aster
to it and see if you get a connection
message on the Asterisk CLI. Do it from the machine itself and then
outside. If it works from the machine itself but not outside, then
you've narrowed it down more.
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Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL
for all transports in PJSIP, and when a connection is established it is
logged to the console.
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Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
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). Try to isolate things further, start from Asterisk itself.
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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
p set logger
on" does the message show up? What is the COMPLETE console output when a
client connects? We have tests which cover TCP and they are working, so
it's likely something environment specific.
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Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville
Sonny Rajagopalan wrote:
Does this help:
Yes, the transport parameter is in the Contact header so it's
interesting it didn't work. If you use pjsip show contacts what is the
contact for the AOR?
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Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL
not have to specify another transport section
with transport=ws?
Correct, you don't.
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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
would be useful.
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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Carlos Chavez wrote:
On 2/15/16 12:50 PM, Joshua Colp wrote:
Carlos Chavez wrote:
Is it possible to use serveral protocols for a single transport section
in pjsip.con? In sip.conf you could use transport=udp,ws,wss so you
cound use webrtc along with your phones but if I try:
[transport-udp
transport is for a specific protocol. You can have multiple.
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Joshua Colp
Digium, Inc. | Senior Software Developer
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Check us out at: www.digium.com & www.asterisk
happen if the URI added does not contain ;transport=tcp which
informs things to use TCP. If the device registering doesn't do this
then it will try to use a UDP transport instead, if not available then
it will fail.
What is the REGISTER from the device?
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Joshua Colp
Digium, Inc. | Senior Software Dev
://issues.asterisk.org/jira
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
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s a result of this it may change but if you use the
tags the information will be there.
To get 13.7.0 you would use:
https://github.com/asterisk/asterisk/blob/13.7.0/asterisk-13.7.0-summary.html
And for 13.7.1:
https://github.com/asterisk/asterisk/blob/13.7.1/asterisk-13.7.1-summary.html
Cheers,
--
J
and transports aren't
found then this would be an issue, which would need console output and
configuration.
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Joshua Colp
Digium, Inc. | Senior Software Developer
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Check us out at: www.digium.com & www.asterisk
at
startup across the board?
Transports can only be loaded at startup. This stems from PJSIP not
being dynamic with transports (it doesn't like its environment changed
to that degree while in use). I'm afraid if your IP changes you'd have
to restart Asterisk when you are using PJSIP.
--
Joshua Colp
James Cloos wrote:
"JC" == Joshua Colp<jc...@digium.com> writes:
JC> This stems from PJSIP not being dynamic with transports (it
JC> doesn't like its environment changed to that degree while
JC> in use). I'm afraid if your IP changes you'd have to restart
JC>
ideas you can offer.
Bryant
The res_pjsip module does not currently support an auto-updating
mechanism for the external signaling and media address information.
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Check us out
the table and forcing a reload
each time an IP address changed might a workable solution.
No, once loaded the transports can not be changed.
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Joshua Colp
Digium, Inc. | Senior Software Developer
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Check us out at: www.digium.com & www.asterisk
We tried this but it seemed that the [asterisk-1] section in pjsip.conf
had no effect. Our sorcery.conf is attached.
Is this possible, and how do we do it? Thanks very much for any advice.
It's not possible to do this. Each source (realtime, config file)
provides the complete definition.
--
Joshua C
IN A 81.23.228.129
proxy.sipthor.net. 60 IN A 85.17.186.7
proxy.sipthor.net. 60 IN A 81.23.228.150
Cheers,
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Joshua Colp
Digium, Inc. | Senior Software Developer
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Check us out at: www.digium.com
it instead or if you are attempting to read
it PJSIP uses a different method (PJSIP_HEADER dialplan function).
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
cause this to happen. Only contacts were
mentioned but it would impact other things. There is now an rc3 which
has a fix for this in it.
[1] https://issues.asterisk.org/jira/browse/ASTERISK-25689
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 358
. Instead, it is passing
companyname.com.
The domain in the From header can be set using from_domain on the endpoint.
It would also be useful to provide the configuration, minus passwords,
and an example of a working SIP INVITE from another machine...
--
Joshua Colp
Digium, Inc. | Senior
Dan Cropp wrote:
outbound_proxy = chi-sbc3-iad.bluip.com
Try setting this to: outbound_proxy = chi-sbc3-iad.bluip.com\;lr
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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
idge from ARI with an id of one that already exists.
[1] http://lists.digium.com/cgi-bin/mailman/listinfo/asterisk-app-dev
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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
from
ARI[1]. Specifically have an ARI operation to redirect a channel into a
Stasis application or to elsewhere in the dialplan. I'm sure people
would be happy if it could be done.
[1] https://wiki.asterisk.org/wiki/display/AST/ARI+Feature+Wish-list
--
Joshua Colp
Digium, Inc. | Senior Software
.
--
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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did you build
chan_rtp with Unicast support? Is it in Trunk or SVN testing branch?
It is currently only available in git master, I haven't tried throwing
it into 13 so I'm uncertain of what all would need to be changed.
--
Joshua Colp
Works as expected using git master branch, I'm not running git
of Asterisk did you build
chan_rtp with Unicast support? Is it in Trunk or SVN testing branch?
It is currently only available in git master, I haven't tried throwing
it into 13 so I'm uncertain of what all would need to be changed.
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis
/
--
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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On 15-10-19 09:12 AM, Andrew Colin wrote:
Do you know if this can be achieved with the standard sip stack in asterisk?
If you are referring to chan_sip I don't believe so but it is possible
there is some obscure option or method to do it that I am aware of.
--
Joshua Colp
Digium, Inc
not currently be modified on a per-endpoint basis
and takes its values from the generated From header. On a global scale
it could be controlled using the default_user global option. Otherwise
there's no real way without adding explicit support for it.
--
Joshua Colp
Digium, Inc. | Senior Software
server may contain that user (or may not,
depending on their configuration/deployment).
I know that in freeswitch there is the option extension-in-contact.
We basically need to achieve the same functionality
It would require modifying the code and adding support.
--
Joshua Colp
Digium, Inc
unlikely that the
problem can be located without exact details for reproducing it. If you
can get a backtrace though you can file an issue on the issue tracker[2].
[1] https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace
[2] https://issues.asterisk.org/jira
--
Joshua Colp
Digium, Inc
with different realms and
such?
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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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of queries per request and that's again extremely inefficient.
The answer to both of your questions is no, there is currently no way to
disable either.
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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
d:
https://issues.asterisk.org/jira/browse/ASTERISK-25439
https://issues.asterisk.org/jira/browse/ASTERISK-25435
https://issues.asterisk.org/jira/browse/ASTERISK-25421
https://issues.asterisk.org/jira/browse/ASTERISK-25378
https://issues.asterisk.org/jira/browse/ASTERISK-25279
They're in the queu
. I greatly regret that moved from
chan_sip to res_pjsip. Previously used very much lacking, and much of
the promise failed. Dmitriy Serov.
Any particular examples?
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Joshua Colp
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Check us out
On 15-10-05 05:58 PM, Dmitriy Serov wrote:
05.10.2015 23:24, Joshua Colp пишет:
On 15-10-05 05:22 PM, Dmitriy Serov wrote:
Hello. Do I understand correctly that the current implementation
res_pjsip does not support ZRTP?
http://lists.digium.com/pipermail/asterisk-dev/2013-December/064401.html
res_pjsip_outbound_registration and have no
outbound registrations it will execute some queries against your
database but otherwise do nothing.
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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
you can
add a comment to an initial page and if your suggestions look good then
Rusty can provide you wiki edit access.
[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
ase
table.
Thanks
Bryant
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On 15-10-04 01:42 PM, Bryant Zimmerman wrote:
*From*: "Joshua Colp" <jc...@digium.com>
*Sent*: Sunday, October 4, 2015 12:12 PM
*To*: asterisk-users@lists.digium.com
*Subject*: Re: [asterisk-users
Outbound registrations are done in res_pjsip_outbound_registration, as a
result the registration= needs to be in a section for that module instead.
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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 15-09-28 09:16 AM, Emil Ohlsson wrote:
Sorry for the delay here. For some reason the mail from Joshua Colp
failed to deliver to my mailbox.
So, anyway, I've set up a local scenario on my computer a PJSIP
client and Asterisk 11.17.1 (On a fedora linux workstation) with the
settings listed
?
MessageSend has no concept of TLS, it gets passed to chan_sip which then
sends it. It's therefore up to chan_sip to do it. It should work.
Haven't done it though.
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Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
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pjsip and
then asterisk.
Same problem. Asterisk was built against static instead of shared.
What's the output of:
ls -al /usr/lib/libpj*
pkg-config --libs libpjproject
And from the pjproject source directory:
grep "aconfigure" config.status
--
Joshua Colp
Digium, Inc. | Senio
On 15-09-24 08:45 AM, Ryan, Travis wrote:
I could bet it could be something in pkg-config. How do I get that to reset? I
never changed anything in there.
It is an installed file from pjproject.
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Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL
file is specifying /usr/local/lib - are there any
remaining libpj files in there?
ls -al /usr/local/lib/libpj*.so
ls -al /usr/local/lib/libpj*.a
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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
of Asterisk may work.
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Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
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is what Asterisk
uses.
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on
https://wiki.asterisk.org/wiki/display/AST/Setting+up+PJSIP+Realtime#SettingupPJSIPRealtime-InstallingandUsingAlembic
will cause alembic to upgrade the tables.
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Joshua Colp
Digium, Inc. | Senior Software Developer
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On 15-09-24 06:30 PM, Ryan, Travis wrote:
Yes, the schema can change between versions. Following the
instructions on
https://wiki.asterisk.org/wiki/display/AST/Setting+up+PJSIP+Realtime#SettingupPJSIPRealtime-InstallingandUsingAlembic
will cause alembic to upgrade the tables.
-- Joshua Colp
--with-external-speex --with-external-gsm --with-external-srtp
--disable-sound --disable-resample --prefix=/usr
Those are the options I use normally on Ubuntu 14.04.
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Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
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installed? That is what the build
system uses for finding the right stuff.
Also what does the following show:
ls /usr/lib/libpj*.so
ls /usr/local/lib/libpj*.so
And what did you pass to the configure script for pjproject?
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW
On 15-09-23 12:14 PM, Ryan, Travis wrote:
Spoke too soon. Same thing.
Josh, any other ideas?
Not really, that's the exact configure line I use. You may have to do a
"make distclean" on both pjproject and asterisk.
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan D
are on the wiki[1] for getting a backtrace. As well - what
version of Asterisk?
[1] https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace
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Joshua Colp
Digium, Inc. | Senior Software Developer
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Asterisk modules installed and did not remove them before installing the
new Asterisk. Essentially res_hep_pjsip is using a self contained copy
of PJSIP which is in an undefined state.
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Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check
On 15-09-23 07:36 PM, Ryan, Travis wrote:
I've got the backtrace, but how much of the info do you want?
Ideally everything.
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Check us out at: www.digium.com & www.asterisk
remaining libraries.
You should also ensure old Asterisk modules don't exist by doing rm -rf
/usr/lib/asterisk/modules
And re-running make install in Asterisk afterwards
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Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
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and send a message out of dialog.
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445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org
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only[1] and has
not received bug fixes for quite some time. It will also go end of life
at the end of this year.
[1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us
offers to this list. They belong on the
asterisk-biz mailing list.
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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
an existing
stream. Using wireshark I can see that the TLS stream is still alive
as there is no tear down communication.
What is the complete console output as well as endpoint configuration?
It should work fine, provided stuff is configured and used right.
--
Joshua Colp
Digium, Inc. | Senior
not work.
Because they both sms and call are coming to the same context
'from-internal', as I notice. I wonder how could i custom the context
for the messaging ?
As I originally mentioned the "message_context" option can be used to
send messages to a different context.
--
Joshua Colp
D
-endpoint_message_context
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Joshua Colp
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+Realtime
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.
There is an issue open[1] to make that configurable but noone has done
it as of this time.
[1] https://issues.asterisk.org/jira/browse/ASTERISK-24106
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Joshua Colp
Digium, Inc. | Senior Software Developer
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console output and do you have an indications.conf
configuration file?
I ask because in this scenario Asterisk would be generating the ringback
itself as audio.
Cheers,
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Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out
was expected - but I am unsure.
There is no enforced minimum/maximum. The value provided is in dB.
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
for quite a long time. If dialing a peer it's resolved at configuration
time, weight/priority should be sorted, and then the top one used. It's
not re-resolved later.
Cheers,
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out
the
headers to appear?
--
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
--
_
-- Bandwidth and Colocation Provided by http://www.api
public phones, and I am in a minority situation where
I am talking to a mix of setups.
Most people run without direct media unless they know the network
topology will allow it 100%.
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check
count for some reason.
snip
This is actually enforced by the system, not by Asterisk itself.
--
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
ultimately and either
that is incorrect for some reason or something is blocking 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
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 221,688 changed lines.
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive
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
--
_
-- Bandwidth and Colocation
occurring.
The call-limit configuration option is what prevents the calls, and it
has no separation between inbound and outbound. You would have to use
other constructs (such as group counting in the dialplan) to do it.
Cheers,
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis
a packet capture, look at the exchange in Wireshark, and
see how the negotiation flows. It requires a basic understanding of ICE.
--
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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