search for: external_signaling_port

You need to put your external IP in the transport configuration: external_media_address=X.X.X.X external_signaling_address=X.X.X.X external_signaling_port=5060 On 21/06/23 12:36, TTT wrote: > I've split this thread off from another (PJSIP authentication) because I think the root cause is something different. I think the problem is the following FROM line in my SIP INVITE transaction: > > From: "MYNAME" <sip:16667778888...

...ement=5 threadpool_idle_timeout=0 threadpool_max_size=100 [transport-udp] type                       = transport symmetric_transport        = yes protocol                   = udp bind                       = 0.0.0.0:5060 external_media_address     = 152.X.Y.Z external_signaling_address = 152.X.Y.Z external_signaling_port    = 5060 allow_reload               = no tos                        = cs3 cos                        = 3 local_net                  = 127.0.0.1/24 local_net                  = 192.168.50.0/24 local_net                  = 192.168.1.0/24 local_net                  = 10.3.2.0/24 local_net            ...

I've split this thread off from another (PJSIP authentication) because I think the root cause is something different. I think the problem is the following FROM line in my SIP INVITE transaction: From: "MYNAME" <sip:16667778888 at 172.31.253.4>;tag=773a3e6a-a677-4fb1-95fc-54b379b650a4 The IP address above is an internal/non-routable IP, so Twilio is rejecting it. For some

...mes out and sends another INVITE, this time with port 1235. Asterisk sends the Trying to port 1235 Third party doesn't monitor this port so it eventually times out and sends another INVITE, this time with port 1236 Asterisk sends the Trying to port 1236 Would this be achieved via the transport external_signaling_port setting? Have a great day! Dan -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Joshua Colp Sent: Tuesday, July 14, 2015 2:04 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [...

...: 1 bind : 0.0.0.0:5061 ca_list_file : cert_file : /etc/asterisk/sslcert.pem cipher : cos : 0 domain : external_media_address : external_signaling_address : external_signaling_port : 0 local_net : method : tlsv1 password : priv_key_file : protocol : tls require_client_cert : No tos : CS0 verify_client : No verify_server...

I am currently running Asterisk 13.1.0-1 I have a chan_sip configuration that works fine with a 3rd party. Third party does not use authentication or registration, it's ip based authentication... When I try switching to PJSIP.conf, I seeing 488 responses from the Asterisk side. What has me really baffled is the debugging indicates [Jul 14 17:28:24] DEBUG[3620] pjsip: sip_endpoint.c

I'm quite new to Asterisk and using Asterisk 13 on FreeBSD current. Asterisk is behind a NAT router, the physical setup is very much a trivial one. The Asterisk PBX is supposed to act as the telephone gateway for several VoIP/SIP phones. I'm using throughout pjsip as configuration, I have no experience with chan_sip since I started recently using Asterisk for several SoHo and lab's