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roberto.medola at gasp...
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 Posted: Mon Aug 17, 2020 4:16 pm Post subject: [asterisk-users] Queue don't call Interface PJSIP |
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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 Called PJSIP/6001, but the extension doesn't ring. If I call from extension to extension, it works normally.
telenet:
Action: QueueAdd
Queue: queuetest
MemberName: 1234
Interface: PJSIP/6001
StateInterface: PJSIP/6001
Ringinuse: yes
Paused: false
If I change to SIP, the extension will call normally.
My configuration pjsip.conf
[transport-udp-nat]
type=transport
protocol=udp
bind=0.0.0.0:5160
local_net=192.0.0.0/24
external_media_address=192.168.0.196
external_signaling_address=192.168.0.196
[6001]
type=endpoint
context=callcenter
disallow=all
allow=g729
allow=ulaw
allow=gsm
auth=6001
aors=6001
transport=transport-udp-nat
direct_media=no
allow_subscribe=yes
sub_min_expiry=30
[6001]
type=auth
auth_type=userpass
password=6001
username=6001
[6001]
type=aor
max_contacts=99
Has anyone experienced the same problem? I upgraded my asterisk to 17 and the problem still persists.
Thanks! |
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jcolp at sangoma.com
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| Posted: Mon Aug 17, 2020 4:58 pm Post subject: [asterisk-users] Queue don't call Interface PJSIP |
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On Mon, Aug 17, 2020 at 6:16 PM Roberto <roberto.medola@gasparimsantos.com.br (roberto.medola@gasparimsantos.com.br)> wrote:
| Quote: |
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 Called PJSIP/6001, but the extension doesn't ring. If I call from extension to extension, it works normally.
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Can you describe the actual network setup further? Is the endpoint behind NAT or merely Asterisk? I ask because there is no NAT configuration for the endpoint, which if it is behind one can be problematic. Failing that you'll need to provide a SIP trace using "pjsip set logger on" to show the actual SIP traffic flowing (and where to).
--
Joshua C. Colp
Asterisk Technical Lead
Sangoma Technologies
Check us out at www.sangoma.com and www.asterisk.org |
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roberto.medola at gasp...
Guest
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| Posted: Tue Aug 18, 2020 7:01 am Post subject: [asterisk-users] Queue don't call Interface PJSIP |
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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 testing is to replace SIP with PJSIP on this client. But as the queue is unable to call a PJSIP extension, the migration project on the client is stopped.
I tried to separate the debug file, but it seems to me that in asterisk 17.16.0, there is a problem or I did not know how to configure it, because the log did not generate it either.
on console:
"pjsip set logger on"
"pjsip set history on"
on file Logger.conf:
debbuger => debug, trace
asterisk -rx "reload"
Make same calls, and opening the file only the following appears:
[2020-08-18 08:46:47.778] Asterisk 17.6.0 built by root @ asterisk-homolog on a x86_64 running Linux on 2020-08-13 22:40:11 UTC\
Em 17/08/2020 18:57, Joshua C. Colp escreveu:
| Quote: | On Mon, Aug 17, 2020 at 6:16 PM Roberto <roberto.medola@gasparimsantos.com.br (roberto.medola@gasparimsantos.com.br)> wrote:
| Quote: |
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 Called PJSIP/6001, but the extension doesn't ring. If I call from extension to extension, it works normally.
|
Can you describe the actual network setup further? Is the endpoint behind NAT or merely Asterisk? I ask because there is no NAT configuration for the endpoint, which if it is behind one can be problematic. Failing that you'll need to provide a SIP trace using "pjsip set logger on" to show the actual SIP traffic flowing (and where to).
--
Joshua C. Colp
Asterisk Technical Lead
Sangoma Technologies
Check us out at www.sangoma.com and www.asterisk.org
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jcolp at sangoma.com
Guest
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| Posted: Tue Aug 18, 2020 7:08 am Post subject: [asterisk-users] Queue don't call Interface PJSIP |
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On Tue, Aug 18, 2020 at 9:00 AM Roberto <roberto.medola@gasparimsantos.com.br (roberto.medola@gasparimsantos.com.br)> wrote:
| Quote: | 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 testing is to replace SIP with PJSIP on this client. But as the queue is unable to call a PJSIP extension, the migration project on the client is stopped.
I tried to separate the debug file, but it seems to me that in asterisk 17.16.0, there is a problem or I did not know how to configure it, because the log did not generate it either.
on console:
"pjsip set logger on"
"pjsip set history on"
on file Logger.conf:
debbuger => debug, trace
asterisk -rx "reload"
Make same calls, and opening the file only the following appears:
[2020-08-18 08:46:47.778] Asterisk 17.6.0 built by root @ asterisk-homolog on a x86_64 running Linux on 2020-08-13 22:40:11 UTC\
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The PJSIP packet logging are verbose messages, if verbose is enabled on console or file they will show up there. The history module also uses CLI commands to examine the history log.
--
Joshua C. Colp
Asterisk Technical Lead
Sangoma Technologies
Check us out at www.sangoma.com and www.asterisk.org |
|
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roberto.medola at gasp...
Guest
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| Posted: Tue Aug 18, 2020 7:57 pm Post subject: [asterisk-users] Queue don't call Interface PJSIP |
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[SOLVED]!!!
My function that changed the callerid was returning an invalid number. Although the asterisk sends the call, the SIP header was wrong and the extension did not ring
Thanks.
Em 18/08/2020 09:07, Joshua C. Colp escreveu:
| Quote: | On Tue, Aug 18, 2020 at 9:00 AM Roberto <roberto.medola@gasparimsantos.com.br (roberto.medola@gasparimsantos.com.br)> wrote:
| Quote: | 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 testing is to replace SIP with PJSIP on this client. But as the queue is unable to call a PJSIP extension, the migration project on the client is stopped.
I tried to separate the debug file, but it seems to me that in asterisk 17.16.0, there is a problem or I did not know how to configure it, because the log did not generate it either.
on console:
"pjsip set logger on"
"pjsip set history on"
on file Logger.conf:
debbuger => debug, trace
asterisk -rx "reload"
Make same calls, and opening the file only the following appears:
[2020-08-18 08:46:47.778] Asterisk 17.6.0 built by root @ asterisk-homolog on a x86_64 running Linux on 2020-08-13 22:40:11 UTC\
|
The PJSIP packet logging are verbose messages, if verbose is enabled on console or file they will show up there. The history module also uses CLI commands to examine the history log.
--
Joshua C. Colp
Asterisk Technical Lead
Sangoma Technologies
Check us out at www.sangoma.com and www.asterisk.org
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