Displaying 20 results from an estimated 5000 matches similar to: "Queue don't call Interface PJSIP"
2020 Aug 18
2
Queue don't call Interface PJSIP
Hi Joshua, thanks for answer.
In this particular test my extension is on a simple network. There is no
NAT, just an asterisk running on a virtual machine on a 7- 64bit CentOs.
I am simulating an environment to be able to use PJSIP on my client. And
even in this small environment, my extension does not call.
My problem with NAT was with SIP "one way audio" on a client. All of
this
2020 Sep 22
3
Asterisk Drop call
Hello.
Thanks for the reply.
Yes. In the traffic analyzed, the BYE is sent by the originator of the
call, but there is no "human" hangup, but the asterisk one.
BYE is sent, received and confirmed.
I don't know how I could investigate the reason for this BYE.
Em 21/09/2020 17:12, Dovid Bender escreveu:
> Is there anything in the Asterisk logs? Which side sends the BYE? Were
2020 Aug 17
0
Queue don't call Interface PJSIP
On Mon, Aug 17, 2020 at 6:16 PM Roberto <
roberto.medola at gasparimsantos.com.br> wrote:
> Hello.
>
>
> I am having a lot of problems with SIP through NAT. So, I decided to adopt
> PJSIP. However, I am not able to make the extensions ring when receiving a
> call from the queue. I'm using telnet to include the extension and on the
> asterisk console, it even shows
2020 Sep 21
2
Asterisk Drop call
Hello
I have an asterisk 16.2.1 on an ubuntu on AWS, which is experiencing a
drop in call. It does not have a certain time, it is random. The audio
is flowing normally and the call is dropped.
Has anyone ever experienced this?
My settings changed below:
allowoverlap = no
udpbindaddr = 0.0.0.0
tcpenable = no
tcpbindaddr = 0.0.0.0
transport = udp, ws, wss
srvlookup = yes
directmedia = no
2020 Aug 18
0
Queue don't call Interface PJSIP
On Tue, Aug 18, 2020 at 9:00 AM Roberto <
roberto.medola at gasparimsantos.com.br> wrote:
> Hi Joshua, thanks for answer.
> In this particular test my extension is on a simple network. There is no
> NAT, just an asterisk running on a virtual machine on a 7- 64bit CentOs. I
> am simulating an environment to be able to use PJSIP on my client. And even
> in this small
2015 Jan 08
2
Asterisk 13.1.0/PJSIP peer IP address issue
I am following the instructions in
https://wiki.asterisk.org/wiki/display/AST/Basic+PBX+Functionality and I am
trying to make a call from extension Alice (6001) to extension for Bob
(6002). When I make the call, I can hear the ringing on Alice's phone
(caller), but Bob's phone (callee) doesn't ring, or show a call coming in
from Alice. My setup and environment is as follows: Alice, Bob
2015 Jul 13
3
How to dial extensions asynchronous-sequentially ?
Hi.
I my dialplan I have :
same = n,Dial(PJSIP/6001,10)
same = n,Dial(PJSIP/6002,30)
same = n,Hangup()
The extension 6002 will not be invited until the called party 6001 hangs up or until 10 seconds if nobody answers the call in 6001.
How to call 6001 and immediately call 6002, having 2 phones ringing at same time, but without doing something like this : same =
2015 Jan 08
4
Asterisk 13.1.0/PJSIP peer IP address issue
Thank you for your note, Scott.
I set rewrite_contact=yes for both contacts, and I also had to do
remove_existing=yes because I had to remove the existing contact
information (max_contacts = 1 was preventing new contact information)
using pjsip
qualify demo-alice etc., after which the right IP addresses showed in pjsip
show endpoints. Anyway, it works as expected now, I think. My pjsip.conf is
2015 Jul 13
3
RES: How to dial extensions asynchronous-sequentially ?
Hi SamyGo.
Thank you for the replay. So, let me explain it better:
I knew that I could use something like " same = n,Dial(PJSIP/6001&PJSIP/6002) ".
While every extension (called phones) rings and before anyone answers, SIP 183 messages will be sent to Asterisk from callees. If a called phone answer, the others will be hanged up. It is ok for me. I want to connect the caller just
2013 Sep 24
1
PJSIP Identify Wiky
The Wiky needs to be updated
https://wiki.asterisk.org/wiki/display/AST/Configuring+res_pjsip#Configuringres_pjsip-IDENTIFY%28res_pjsip_endpoint_identifier_ip%29
This is the example shown:
"[6001]
endpoint=6001
match=203.0.113.1"
It should be:
"[6001]
type=identify
endpoint=6001
match=203.0.113.1"
2016 Mar 03
3
RTP / NAT question ( pjsip )
Thank you for the response Joshua .
I had rtp_symmetric=yes before I wrote the email, then I set it to no, restart asterisk, and tried to make the call from the remote endpoint again but still tcpdump is showing me the RTP packets are being sent from Asterisk to the private IP.
tcpdump on asterisk server showing UDP packet bound for my remote endpoints internal IP:
17:07:57.130212 IP
2016 Jan 18
2
How to get PJSIP SIP messages in a log file and not in console ?
Hello,
How should I configure Asterisk (13.7.0) to get persistently PJSIP SIP
messages in a log file and not in console ?
I would expect adding "debug=yes" in pjsip.conf to produce the same output
as "pjsip set logger on".
Am I understanding correctly ?
Best regards
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2017 Sep 14
2
Realtime pjsip issues
We are having an issue where on the latest version of asterisk when
configuration pjsip via realtime.
we do a pjsip list endpoints it shows our endpoints but lists them as
invalid.
When we do the pjsip list endpoints again it shows no objects.
This applies to pjsip list aors as well. We did not have this issue on
our older asterisk 13 installs. My guess is something has changed
2016 Jul 02
3
Registration server with PJSIP
Hello,
I am moving from realtime chan_sip to pjsip and one of the problem I am
facing is the lack of "sipregs". With chan_sip, when an extension
registers, the server where it has registered to is stored in sipregs.
Is there something similar in pjsip? How can I find on which server the
pjsip extension has registered to?
Leandro
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2017 Oct 21
2
PJSIP trunk to Telynx
Has anyone used Telynx as a SIP trunk provider?? It works with chan_sip
but it I seem to be having problems trying to set up a PJSIP trunk.? I
always get a 401 Unauthorized when they send me a call.? I know my
username and password are correct since I can register and PJSIP uses
the same information for inbound as for the registration.? Unfortunately
their support department said "PJSIP
2017 Nov 19
2
pjsip subscribe (presence) always returns: No matching endpoint found
Hello List
I am in the progress of migrating from chan_sip to pjsip.
I fear I have missed something on how hints need to be specified for
pjsip.
For chan_sip I have configured sip.conf
subscribecontext = localuser
and in the dialplan I set:
[localuser]
exten => 11,hint,SIP/11
Now if a phone subscribes to '11' this works.
Now I try to get the same working for pjsip. I understood
2016 Nov 04
2
pjsip transports from database.
Hey all
I am trying to configure all my pjsip transports form a database table.
The issue I am running into is that pjsip is auto binding to 0.0.0.0:5060
before it reads my list of transports from the database. This means that my
entries for port 5060 are already bound and the settings in the database
are not loaded.
When loading the transport form the .conf file it works as expected
2017 Jun 11
2
asterisk 13.16 / pjsip / t.38: res_pjsip_t38.c:207 t38_automatic_reject: Automatically rejecting T.38 request on channel 'PJSIP/91-00000007'
On Sun, Jun 11, 2017, at 11:34 AM, Michael Maier wrote:
<snip>
> >
> > PJSIP uses a dispatch model. The request is queued up, acted on, and
> > then that's it. The act of acting on it removes it from the queue.
>
> That's the *expected* behavior ... . I rechecked again and again. All
> existing tcpdumps. The "resent" package isn't part of
2015 Jul 13
2
RES: RES: How to dial extensions asynchronous-sequentially ?
Hi Sammy.
After answering your last message (please, see my last message), I was thinking about conferences and my main objective.
Conferences will not work well for my case, because I it will allows more than one called party answering the call. But, after one answers the call, I need cancel the others ringing callees.
In this case, maybe the best thing to do is to let the called party sends
2014 Sep 05
2
Asterisk with PJSIP
Hi All,
I installed Astreisk 13beta with pjproject 2.3(2.2.1) from source code on
CentOS7.
--
https://wiki.asterisk.org/wiki/display/AST/Building+and+Installing+pjproject
The installation is OK.
But the connected SIP cilents (both Linphone on Windows7) cannot
communicate.
I hope your comment such as the testing for resolving the problem.
My status is the following(1 and 2).
Why 'Everyone