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! -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20200817/5f032311/attachment.html>
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 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20200817/7bb689e2/attachment.html>
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:> On Mon, Aug 17, 2020 at 6:16 PM Roberto > <roberto.medola at gasparimsantos.com.br > <mailto: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 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 <http://www.sangoma.com> and > www.asterisk.org <http://www.asterisk.org> >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20200818/a4296248/attachment.html>