with those SIP devices.
>
> 1. Is it possible ? I can use any Asterisk version for implementation.
It is not possible to configure Asterisk for this. The chan_pjsip module only
does normal reinvites with SDP when configured to pass through MOH signaling.
--
Joshua Colp
Digium - A Sangoma Co
seen as an equivalent
> or near-equivalent of an AGI Command...
I looked at the AGI command list and didn't see any that weren't possible in
dialplan where it made sense. Do you have further examples?
--
Joshua Colp
Digium - A Sangoma Company | S
t; without the need to call an external script.
In particular for this it can done in dialplan using the TIMEOUT dialplan
function[1].
[1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Function_TIMEOUT
--
Joshua Colp
Digium - A Sangoma Company | Senior Software Developer
445 Jan
;d be better off using
JsSIP example code instead for making a solution in that area.
--
Joshua 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 the code which introduced this feature and couldn't find
> anything obvious why this is happening.
Have you bumped up the core debug to see what's going on underneath? There will
be information about whether it is really generating the DTMF in the core, and
if so then it'd
On Wed, Sep 12, 2018, at 2:25 PM, sean darcy wrote:
> On 9/12/18 1:22 PM, Joshua Colp wrote:
> > On Wed, Sep 12, 2018, at 2:19 PM, sean darcy wrote:
> >> I understand that HangUp() hangs up the calling channel. I want to
> >> hangup the called channel.
> >>
>
quot;send". How are you doing it?
--
Joshua 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
things.
--
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
thought invites had to go to port 5060 or so. I don't understand why
> somebody (let's assume a bad guy) is trying ports above 5.
There is nothing that explicitly states that it has to be 5060, and in the case
of the above it's just a random source port.
--
Joshua Colp
Digium, I
on this behavior?
Cheers,
[1] https://gerrit.asterisk.org/#/c/asterisk/+/9822/
--
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
strictrtp option. Otherwise it's not something in Asterisk that stops
this kind of stuff, it's the NAT Implementation in the router.
--
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
-
ent doesn't exist anymore[1] in Asterisk 13 and above.
[1]
https://blogs.asterisk.org/2016/02/10/converting-from-chan_agent-to-app_agent_pool/
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out a
On Sat, Jul 28, 2018, at 6:28 PM, Jonathan H wrote:
> OK, thanks. Shall I file a ticket to get that example file updated?
Sure!
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.aster
007f62a9ba2646 sp 7ffc9215d408 error 4 in
> libc-2.27.so[7f62a9af1000+1e7000]
>
> Took that line back out, and Asterisk started again. Shall I file a bug?
Yes, issues should be filed on the issue tracker[1]. It may be something
particular about your config.
[1] https://issues.as
romsip2sip
> aor/max_contacts = 3
> registration/contact_user = myusername
> outbound_proxy = proxy.sipthor.net
> endpoint/language=en_GB
This is an ITSP trunk, you've configured it kind of as if it were a phone.
Instead of "accepts_registrations" you li
address resolve a hostname down to all addresses (including
SRV) - not just a single one.
Outgoing supports A, , SRV, and NAPTR automatically.
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.
e found it simple! It was a challenge to strike the right balance
between the interface and giving full power over what you can do but I've found
once it finally clicks people generally go "that makes sense".
--
Joshua Colp
Digium, Inc. | Senior Software Developer
44
and add the snoop channel and the channel doing the
snooping into it.
--
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
___
asterisk-app
of any current issues for this and haven't seen any. I'd suggest
filing an issue[1] with a backtrace so it can be narrowed down. It may be
something particular to your usage that noone else has seen.
[1] https://issues.asterisk.org/jira
--
Joshua Colp
Digium, Inc. | Senior
nism to do otherwise,
outside of somehow creating something yourself.
--
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
--
no feature added to
control it. Someone would need to go into the code, define what needs to happen
and how it can be controlled, and implement it (in regards to MixMonitor and
ChanSpy).
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
; >
> >
> > Ordering by includes works for me under Asterisk 11 and 13
The context always takes priority over includes. Includes are only examined if
there are no matches in the current context. It's always worked this way.
Ordering includes as such is one way to control that
place being
one of the people who respond there :P.
[1] https://community.asterisk.org/
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com &
On Fri, May 11, 2018, at 10:36 AM, Steve Edwards wrote:
> On Fri, 11 May 2018, Joshua Colp wrote:
>
> >> In the above example, even though the INVITE/SDP says they prefer gsm
> >> over ulaw and the OK/SDP says I prefer ulaw over gsm, they can choose
> >>
in devices won't allow it - they require a single codec be in use for each
direction.
--
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
--
___
ound package? We are
> considering to support Hebrew and possibly Yiddish.
The actual submission process including what is required is documented on the
wiki[1].
[1]
https://wiki.asterisk.org/wiki/display/AST/Asterisk+Sounds+Submission+Process
--
Joshua Colp
Digium, Inc. | Senior Software Develo
configuring instead of having
things just try to figure out what is in use.
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com &
will always strip and regenerate the DTMF
tone. You have to purposely misconfigure things to cause it to not get
stripped. IE: DTMF is actually inband but you configure it for RFC2833. Since
Asterisk wouldn't be listening to the audio stream, it would go right through
and get recorded.
--
username,auth_username Way(s) for Endpoint to be
> identified
The wiki documentation hasn't been regenerated lately (it's in queue to be
fixed). "username,auth_username" would be correct. There's also others[1]
depending on version.
[1]
https://github.co
both at IP settings,
> ignoring From header for identification but using it for other things
> (setting CallerID, ...).
It depends on configuration, but ultimately it can only be identified using a
single endpoint identifier - so not in combination, thus by From OR IP.
--
Joshua Colp
Digi
ith a packaged Asterisk 13.14.1, I edited
> a pjsip.conf file with the following content (and nothing more):
Your version is also quite old, and changes/improvements/tweaks have been made
since then to the option.
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW -
ith a packaged Asterisk 13.14.1, I edited
> a pjsip.conf file with the following content (and nothing more):
> [global]
> endpoint_identifier_order=auth_username,ip,username
> max_forwards=50
This is incomplete. You need to also have "type=global".
--
Joshua Colp
Digium, In
24 20:39:21] DEBUG[19041] http.c: HTTP keeping session open.
> status_code:404
>
> Suggestions?
Is there anything in the console at startup stating that stuff didn't load? The
module which does websockets is res_http_websocket, and you can see if all that
is needed is loaded using
On Fri, Apr 13, 2018, at 6:09 PM, Andrzej Nowrot wrote:
> Hi
>
> Is there a way to disable blind and attended transfer during a call.
No, DTMF features are not call time configurable. They are only grabbed when
the channel is first bridged, not as they are potentially used.
Cheers,
-
ffic to get a NAT Binding for incoming
> RTP Early media traffic?
The code is in res_rtp_asterisk[1]. It's not complex and despite the comment is
not specific to video. Without logs showing where things are coming from and
going I don't really have anything else to add.
[1]
https:/
On Mon, Apr 16, 2018, at 12:47 PM, Administrator TOOTAI wrote:
> Le 16/04/2018 à 16:52, Joshua Colp a écrit :
> > On Mon, Apr 16, 2018, at 11:47 AM, Administrator TOOTAI wrote:
> >> Hi all,
> >>
> >> we are trying to move our servers from chan_sip to chan_pjsip
o auth credentials for realm(s)
> 'asterisk' in challenge.
The remote side challenged for authentication but your endpoint has no
"outbound_auth" configured, so chan_pjsip has no idea of how to authenticate.
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Dav
There may be ways to sort of do such things, like listening to an AMI event for
a successful inbound registration, updating configuration, reloading it, and
causing an outbound registration to get sent. It's hackish at best.
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Ja
ehavior - we'll send to the IP address and port they
told us. There's nothing that Asterisk itself can do in that instance, the
endpoint has to send media or place the correct IP address and port in the
messages.
Was any media received from it?
--
Joshua Colp
Digium, Inc. | Senior S
patch forwards video while in an early media state before the call is
answered and bridged, yes.
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com &
> but the recipent phone doesn't get any video from the Asterisk before the
> call.
Ah yeah video, I forgot that it was a recent change to add support for it[1].
It's not yet in any release.
[1] https://gerrit.asterisk.org/#/c/8398/
--
Joshua Colp
Digium, Inc. | Senior Software Deve
where they've told us to send it, which as I've mentioned
can be wrong.
--
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
--
y not have the correct target of
media.
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
--
_
-- Bandwidt
=sip:sip.flowroute.com:5060
Is there any reason you aren't just using sip:flowroute.com here?
PJSIP does SRV resolution so that'll use SRV instead which I know works.
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check u
ssed some other RFC more clear about that topic.
>
> To try to reproduce the problem with our SBC, is there a way to tell
> the asterisk, preferably PJSIP, to directly answer with 180 ringing
> without prior 100 trying?
The PJSIP channel driver has no option or ability to do this. I do
header?
> Is there something else I am missing to perform this?
>
> Have a great day!
Contact is never used for callerid. The only option available is contact_user
on the endpoint to change the Contact username, that's it.
--
Joshua Colp
Digium, Inc. | Senior Software Developer
44
d the "parkedcalls
> show" command no longer exists. So my question is, how do I get that
> information with Asterisk 13?
There is still a CLI command to inspect the parking lot. It's "parking show
".
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan
gt; 172.20.xx.xx:60640
>
> on new Asterisk 15.2 i decide to move to PJSIP but this functionality
> don't work and, on REFER, call dropped.
>
> Maybe there's something needs to be enabled or checked ?
I don't understand the specific scenario here you are referring t
not experienced a problem like this one.
This has been fixed[1] in the branch and will be in the next normal release.
You can pull down the minor change from the review if you want. It tells PJSIP
not to build with support for that.
[1] https://gerrit.asterisk.org/#/c/8193/
--
Joshua Colp
Digi
o". If you really don't want it you can change it
to "no" in sip.conf
--
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
--
__
origination[2] to
send a message using the dialplan[3].
[1]
https://wiki.asterisk.org/wiki/display/AST/Asterisk+15+Endpoints+REST+API#Asterisk15EndpointsRESTAPI-sendMessageToEndpoint
[2] https://wiki.asterisk.org/wiki/display/AST/Asterisk+Call+Files
[3]
https://wiki.asterisk.org/wiki/display/AST
issue?
You'll have to be more specific. Where do you see the %23? In SIP? As the
extension trying to be executed in the dialplan?
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.co
15665' from 10.10.0.8:5060
>
> Why not loolking up "pstn-" in sip.conf?
It found pstn- using 10.10.0.8:5060 - if the request always comes from the
same IP address and port it has no other way built in to differentiate between
the two except by matching based on
On Thu, Feb 15, 2018, at 7:03 PM, the...@sys-concept.com wrote:
> On 02/15/2018 03:44 PM, Joshua Colp wrote:
> > On Thu, Feb 15, 2018, at 6:43 PM, the...@sys-concept.com wrote:
> >> I'm using Audio-codes MP-114 unit and it has two public lines PSTN ports
> >>
"pstn-" (not pstn-9998)
> Where is this label coming from?
It is from the SIP entry in sip.conf that it was matched against.
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asteri
gt;
> I did probably try all possible permutations of:
>
> direct_media=no
> rtp_symmetric=yes
> force_rport=yes
>
> But still no audio.
>
> Any hints on how to force asterisk to send the first rtp packet?
The "rtp_keepalive" option can be used to have the
her (short) delay and process into pipeline. Not a complaint, just a
> question.
I have no timeframe on when such a thing would be done. It's not something that
has been requested before to the best of my knowledge.
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis
out not actually being able to
> *use* Opus once installed, so if anyone can point me to where I've
> gone wrong I'd be most grateful!
The opus support can only be used currently for reading files and for
transcoding (for example one leg in g722 and the other in opus, or for
con
On Fri, Jan 12, 2018, at 3:02 PM, Binarus wrote:
> Thanks for taking the time, but ...
>
> On 12.01.2018 12:04, Joshua Colp wrote:
>
> >> Could this be one of the rare cases where 13 and 15 needed security
> >> fixes, but 14 didn't?
> >
> >
and 15 needed security
> fixes, but 14 didn't?
These are normal bug fix releases, not security releases. As such 14 did not
receive a release.
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davi
the most current if I can.
Can you please file an issue[1] with all the information?
[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
--
__
ut being logged in /var/log/asterisk/
> messages
>
> I would love to have ARI debug log messages in /var/log/asterisk/debug
> or even better in it's own ari-debug file.
That is not something anyone has implemented as of this time. The messages
themselves just get raised as normal
s part of Asterisk 14 work was done in DNS land
(failover to different targets, including between IPv6 and IPv4) and based on
discussions I had with other people at SIPit I made it automatic so that media
family = signaling family. To keep things better in line and to provide a
better e
4 only
> endpoint and is performing an outgoing call, my asterisk server is
> answering with an IP4 RTP IPv6 address:
The rtp_ipv6 option is not needed, in current versions things will
automatically be updated to reflect the signaling. Remove it and give it a try.
The option itself actuall
to change it. Most people end up just doing the parsing in the
dialplan.
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?
"line" support doesn't have an explicit RFC. It relies on the remote
side sending back the contents of the registered Contact address as they
are supposed to when sending the INVITE. In practice some do, some
don't.
--
Joshua Colp
Digium, Inc. | Senior Software Develop
es the request to an endpoint. In the endpoint
if you have no inbound authentication specified (auth option) then it
won't require authentication.
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com
complex workaround?
Within the code f->subclass.integer is where the DTMF digit is. You'd
need to make a code change to set another dialplan variable which
contains it.
[1] https://issues.asterisk.org/jira/browse/ASTERISK-14380
--
Joshua Colp
Digium, Inc. | Senior Software Deve
efine a mechanism to do so and implement
it in the code.
--
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
--
_
han_sip it was just reply 200 OK on keepalive packet without need
> define trunks.
>
>
All incoming traffic into chan_pjsip is matched to an endpoint, this
includes OPTIONS. The OPTIONS request is also treated as if it were an
INVITE per the RFC, which is why the extension also has to
010933a080fe-17271@10.30.100.41) - No matching endpoint found
You would need to add an endpoint for it and have it match, using a
"type=identify" section matching on IP address would work. You would
also need an "s" extension in the context since the OPTIONS request has
not exten
OPTIONS and what is happening now.
--
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 Col
tion, and confirming it was loaded as expected. If it's not then
you can look at the Asterisk console at load time and it will tell you
what it did not like.
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Hun
mportant. When set in an endpoint it
configures unsolicited MWI - that is MWI without the endpoint
subscribing for it. When set in an AOR it configures what mailboxes the
endpoint can subscribe and receive MWI for. Since you've moved it to the
AOR it can now subscribe to the mailbox and
RI that can be used by the remote endpoint to
be provided. The code does not look up an endpoint and try to construct
a SIP URI for you.
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us
e the old
> legacy ipv4 protocol? :-)
Don't specify a transport on the endpoint. Transport selection will
automatically choose the right one in this scenario. The "transport"
option only allows a single transport and it is for forcing a transport
to alway
e intelligent and trying to prevent the phone from
> subscribing to itself?
The chan_pjsip module doesn't prevent that. You'd need to provide the
full SUBSCRIBE now that it is actually finding the endpoint and coming
in.
--
Joshua Colp
Digium, In
:d1d0:204:13ff:fe30:228d:2332' (callid:
> ow21f3eg@snom) - No matching endpoint found
>
> And I in the logger I see that the subscriber request is being rejected
> with error 404.
>
> Any hints what I'm doing wrong?
Have you checked the Asterisk console when PJSIP is loa
t; I should also mention that this is Asterisk version 1.8.12.1
I'm sorry but this version is old enough that what I currently know is
far past it. It may have been possible in that old version for the SSRC
to be as you state. In recent stuff it doesn't seem to be possible.
--
Josh
the
ICE candidates, giving a better chance that things will work.
--
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
--
On Wed, Nov 15, 2017, at 01:05 PM, Carlos Chavez wrote:
> On 11/14/17 5:23 PM, Joshua Colp wrote:
>
> > On Tue, Nov 14, 2017, at 07:19 PM, Carlos Chavez wrote:
> >> Trace with 3 clients. We can hear each other but no video.
> >>
> >> https://pbxoficina
a streams by sending a reinvite to the
participants but we don't get any response, which means for some reason
the browser may not have liked what we provided.
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at:
On Tue, Nov 14, 2017, at 07:03 PM, Carlos Chavez wrote:
> On 11/14/17 4:27 PM, Joshua Colp wrote:
>
> > On Tue, Nov 14, 2017, at 06:25 PM, Carlos Chavez wrote:
> >> On 11/14/17 3:55 PM, Joshua Colp wrote:
> >>
> >>> On Tue, Nov 14, 2017, at 05:47 PM, Car
On Tue, Nov 14, 2017, at 06:25 PM, Carlos Chavez wrote:
> On 11/14/17 3:55 PM, Joshua Colp wrote:
>
> > On Tue, Nov 14, 2017, at 05:47 PM, Carlos Chavez wrote:
> >> I followed the blog post and I can get video from the conference if
> >> I configure the bri
lid though.
Have you confirmed that the maximum number of streams is set using
"pjsip show endpoint"? and that the codecs are correct?
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
s known working configuration.
[1] https://issues.asterisk.org/jira
[2]
http://blogs.asterisk.org/2017/09/20/asterisk-15-multi-stream-media-sfu/
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & ww
mp or wireshark to confirm what
they've said though. I just looked at the code and I don't see a way
that we'd ever have the SSRC be 0.
Cheers,
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Hu
elf to do this already. Logging
to something like Homer might work, or just doing a packet capture.
Otherwise you'd need to make changes to Asterisk to add the
functionality you mention.
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
C
the
asterisk-dev mailing list would be the best place to discuss such things
since that is where developer talk occurs.
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Che
rojects (like those already mentioned) that are a
better fit, and Asterisk can even play a part in there as an application
server.
I'm a firm believer in using the right tool for the right job even if it
means that Asterisk isn't the right fit. Frustrated users are something
I never want to s
ectrtpsetup" equivalent in PJSIP. Even in chan_sip it
was experimental and could break things depending on the codec payloads
in use.
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out
uivalent is this:
[mytrunk]
type=identify
endpoint=mytrunk
match=203.0.113.1
>From the page you linked. That says "Match incoming traffic from
203.0.113.1 and use endpoint mytrunk for it".
You also need an endpoint defined like:
[mytrunk]
type=endpoint
context=from-external
disallo
this is done by using an identify section and matching based on IP
address. There's also the line option[1] to outbound registration which
works with some equipment, if it works then no identify section is
required.
[1]
http://blogs.asterisk.org/2016/01/27/the-pjsip-outbound-registration-line
nk:23] Dial("PJSIP/10-0006",
> "PJSIP/0xx@3x,300,T") in new stack
> -- Called PJSIP/0xx@3x
The PJSIP_HEADER dialplan function operates on the channel it is invoked
on. In this case you are using it on the caller, not the called pa
ainst "13" but it really was against master.
git checkout -b 13 origin/13
Would create a local branch "13" which is from the remote branch "13".
You'll need to do this, or do your "git review" against master and then
cherry pick from inside G
On Tue, Sep 26, 2017, at 05:53 PM, marek cervenka wrote:
> Dne 26/09/2017 v 22:33 Joshua Colp napsal(a):
> > On Tue, Sep 26, 2017, at 05:29 PM, marek cervenka wrote:
> >> hi,
> >>
> >> i want use asterisk+pjsip as voip client with multiple registrations
> >
o only one
> account (because of same ip address/port ?)
>
> how can i specify different source port or different contact address for
> asterisk pjsip client with registration?
The "contact_user" option configures the user portion of the Contact
that is sent in the RE
tion. Does the outbound registration work or not work? Does it
show as registered in PJSIP? If you leave out the "realm" option what
happens? When you say "can't send any calls across the registration"
what does that mean? Are you referring to inbound calls or outbound
ca
de these and me (or another individual) may pick out
what is wrong.
--
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
--
__
as I said
with debug to see what queries are being done to confirm things. You
need to do troubleshooting and isolate things to determine the cause of
the problem. You also did not answer my questions about the database
schema.
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis D
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