similar to: PJSIP Stun/ICE

Joshua Since there is no automated way currently built in to update the external signaling and media address information. Does the realtime pjsip support having the transport contexts section being pulled from a database table? I was thinking a cron script updating the table and forcing a reload each time an IP address changed might a workable solution. Thanks Bryant

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

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

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

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

Hello, Where can I find doc about PJSIP's ice_support parameter ? Do you need to configure things elsewhere in Asterisk config files (rtp.conf, PJSIP transport sections, ...) to make ICE work properly ? I'm asking because, if I'm not mistaken, STUN requires setting a STUN server so I think ICE most probably, should also require settings some public resources. Best regards

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 >>

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

Hi, What is the equal option for externip in asterisk 13 with pjsip. I have tried external_media_address=XX.XX.XX.XX external_signaling_address=XX.XX.XX.XX but asterisk 13 writes local ip to the from header. any suggestions? Best Regards, Madushan -------------- next part -------------- An HTML attachment was scrubbed... URL:

I want to configure communication with my phone provider using TLS for all the obvious reasons. Since I'm behind a firewall, I'll be needing to do it with NAT. There are examples of UDP plus NAT in pjsip.conf, but none for TLS plus NAT. Would it be correct to set up the TLS transport stanza to look like the [transport-udp-nat] stanza example, replacing UDP with TLS in lines like

Bryant, I have the same problem with dynamic public IPs and PJSIP. What is your idea to solve the problem? My suggestion would be to write a script that monitors the change, pjsip.transports.conf updated and Asterisk restarts? Daniel > Am 26.01.2016 um 14:21 schrieb Joshua Colp <jcolp at digium.com>: > > Bryant Zimmerman wrote: >> Joshua >> So once a transport is

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

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 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

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

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

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

On 11/15/17 11:10 AM, Joshua Colp wrote: > On Wed, Nov 15, 2017, at 01:05 PM, Carlos Chavez wrote: >> On 11/14/17 5:23 PM, Joshua Colp wrote: >> >>> On Tue, Nov 14, 2017, at 07:19 PM, Carlos Chavez wrote: >>>> Trace with 3 clients. We can hear each other but no video. >>>> >>>>

>>>>> "AS" == A J Stiles <asterisk_list at earthshod.co.uk> writes: AS> If you are paying for a business-grade Internet connection, you AS> should get a static IP address -- or a block of them -- as AS> standard. Maybe you need to change your ISP? In some places (including here) static ip is not affordable. -JimC -- James Cloos <cloos at