similar to: TLS and NAT

Hello Steve, use the following configuration for the transport and bind this transport to the trunk: [transport_name] type=transport protocol=tls bind=192.168.13.24 ; your bind IP ca_list_file=/etc/pki/tls/certs/ca-bundle.crt ; method=tlsv1_2 verify_server=yes allow_reload=no ;tos=0xb8 ;cos=3 external_media_address=your.ext.host.name ; hostname pointing to your ext. IP

Thanks, Michael. A few questions: Is [transport_name] a reserved word, or am I supposed to replace it with a name of my own, like '[did-transport]'? Some of the keywords I haven't seen before. Is ca_list_file supposed to be an aggregate of the public and private key? And what are the 'method,' 'tos' and 'cos' keywords, which are commented out in your

Greetings ! My goal is to get Twilio trunking working, and with TLS/SRTP. I see this concerning message in my log: [Feb 7 16:50:26] ERROR[20596] res_sorcery_config.c: Could not create an object of type 'endpoint' with id ?twilio' from configuration file ?pjsip.conf? Thus, ?pjsip show endpoints? does not show the endpoint for the Twilio trunk. Hoping for a sanity check of

I've seen this happen three times in the wild now.  I've been trying to isolate the source of the issue, but so far it seems like there's not enough debug output to know why this occurs. Long story short: - Start Asterisk - PJSIP Handles receiving INVITE from ITSP via WAN (Asterisk is behind NAT).  SIP is handled correctly, Asterisk responds OK with RTP media address of

Hey guys,tried to make tls work with pjsip on asterisk 13.2.0 have compiled pjsip with ssl, added transport [tls] type=transport cert_file=/pbx/keys/server.crt ca_list_file=/pbx/keys/ca.key priv_key_file=/pbx/keys/server.key protocol=tls bind=192.168.1.4:5061 local_net=192.168.1.0/24 external_media_address=77.77.77.77 external_signaling_address=77.77.77.77 have configured Grandstream GXP1400

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

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

On Thursday, January 23, 2020 11:31:46 PM CET Sean Bright wrote: > On 1/21/2020 9:18 PM, hw wrote: > > [transport-tls] > > type = transport > > protocol = tls > > bind = 0.0.0.0:5061 > > tos = cs5 > > cert_file = /etc/asterisk/cert/asterisk.pem > > ca_list_file = /etc/pki/tls/certs/ca-bundle.crt > > method = sslv23 > > This is what mine

Hello All, I have an Asterisk server v13.1.0 running on EC2 and I am able to connect and register SIP devices and "see" them on the asterisk CLI. I am also able to place calls, but I am not able to hear any audio on either end after the call is picked up. I was wondering if you can tell me what a minimal configuration for Asterisk on EC2 looks like. My current pjsip.conf configuration

Scenario: Our Asterisk 13 PBX (on network 192.168.254.0/24, bound to 192.168.254.1:5060) is behind a NAT, acting as a client to our ITSPs SIP server. But also, this Asterisk is server for various VoIP telephones. Acoording to Asterisk's wiki, the transport section of pjsip.conf is configured as follows: ; Transport via UDP [transport-nat-udp] type= transport

Ok Dan, try this... I was able to get this to work behind a NAT and with ip address authentication. [global] type = global debug = yes [transport1] type = transport bind = 0.0.0.0 protocol = udp *local_net=<yourlocalnet I.E. 10.10.10.10/24 <http://10.10.10.10/24>>external_media_address=<your public ip address>external_signaling_address=<your public address>*

On Thu, Sep 15, 2016 at 7:17 AM, Faheem Muhammad <faheem2084 at gmail.com> wrote: > > > On Wednesday, 14 September 2016, Madushan Geethanga < > mgliyanage.rc at gmail.com> wrote: > >> Hi, >> >> What is the equal option for externip in asterisk 13 with pjsip. I have >> tried >> >> external_media_address=XX.XX.XX.XX >>

On Thu, Sep 15, 2016 at 8:38 AM, Madushan Geethanga <mgliyanage.rc at gmail.com > wrote: > Hi, > > Thanks for the reply. > > Yes my PABX is on the cloud when I try to register to the server, the > server sends registration OK with public address but OPTION method > includes the private address of the server in from header not the public > address. I have include

On Mon, Dec 15, 2014 at 9:48 PM, Dan Cropp <dan at amtelco.com> wrote: > > Thanks George. > > I will correct my local_net in the morning. > > Vitelity chan_sip settings I have working, do not have a fromuser. > sip.conf settings... > > I think you can actually specify anything, it just has to be populated with something other than a sub-account username. >

On Mon, Dec 15, 2014 at 7:34 PM, Dan Cropp <dan at amtelco.com> wrote: > > I am not sure if I entered the correct settings for the transport > information. > > For the local_net, I entered my local ip address, but no mask. I will > check with the network admin so he can verify the settings I entered. > > > You need the network and mask. For example if the ip

On Fri, Sep 16, 2016 at 5:55 AM, Madushan Geethanga <mgliyanage.rc at gmail.com > wrote: > Hi, > > Tried with both softphone (Ekiga) and snom IP phone, contact header > contains the public IP. but from header contains the private IP. after > OPTIONS method sent by the server. client sends an Register with expires 0. > Ok, did setting from_domain work? > > Best

Hi guys, have really annoying problem with trunks when I calling over voip provider.. after awhile provider sends INFO packages but for some reason Asterisk doesn?t answer on it. after 8 packagers provider just drops the call, here is the package <--- Received SIP request (555 bytes) from UDP:192.168.53.1:5060 ---> INFO sip:c2e74184-4d23-43a3-8fd9-26ba2c5ef6c9 at 192.168.53.9:5060 SIP/2.0

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:

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

Hi, after switching from chan_sip to chan_pjsip, a device running Grandstream Wave leads to the following error message on the asterisk console: SSL SSL_ERROR_SSL (Handshake): Level: 0 err: <336109761> <SSL routines- ssl3_get_client_hello-no shared cipher> len: 0 peer: 10.10.20.29:43357 Something with the encryption must have changed with asterisk. How can I get the device to