Re: [asterisk-users] Trying asterisk on AWS
Thanks for the information This is now working... externip=EC2 public IP localnet=EC2 local range nat=force_rport,comedia I got audio, Fantastic Jerry > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Trying asterisk on AWS
On Thursday 06 October 2022 at 15:24:22, Jerry Geis wrote: > I added: > > externip=xxx > nat=force_rport,comedia > > to the general section of sip.conf > > its still sending to the local IP. Does your local router (the one connecting Linphone to the Internet) have a "SIP helper" or "SIP ALG" feature? If so, ensure that it is turned off. Antony. -- "Tannenbaumschmuck" is a perfectly reasonable German word meaning Christmas tree decorations, and is not a quote from Linus Torvalds. Please reply to the list; please *don't* CC me. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Trying asterisk on AWS
On Thu, Oct 6, 2022 at 10:24 AM Jerry Geis wrote: > >The sample configuration file outlines how things work, and the options for > >it: > >https://github.com/asterisk/asterisk/blob/master/configs/samples/sip.conf.sample#L874 > >in general localnet and externip (or externaddr, or externhost) > > I added: > > externip=xxx > nat=force_rport,comedia > > to the general section of sip.conf > > its still sending to the local IP. > Look at the actual SIP/SDP signaling to see what is being sent and what IP addresses are being used. If they're correct, then see if you are receiving traffic using "rtp set debug on". If you aren't then it's something outside of Asterisk preventing incoming traffic. Until Asterisk receives traffic it can't know the IP address+port to send outgoing to, beyond what was given in the SIP/SDP. -- Joshua C. Colp Asterisk Project Lead Sangoma Technologies Check us out at www.sangoma.com and www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Trying asterisk on AWS
>The sample configuration file outlines how things work, and the options for >it: >https://github.com/asterisk/asterisk/blob/master/configs/samples/sip.conf.sample#L874 >in general localnet and externip (or externaddr, or externhost) I added: externip=xxx nat=force_rport,comedia to the general section of sip.conf its still sending to the local IP. Jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Trying asterisk on AWS
On Thu, Oct 6, 2022 at 10:17 AM Jerry Geis wrote: > > > On Thu, Oct 6, 2022 at 9:02 AM Jerry Geis wrote: > >> >> >> On Thu, Oct 6, 2022 at 8:59 AM Jerry Geis wrote: >> >>> I am trying to get audio to work on AWS using asterisk 18.14.0 >>> >>> I have enabled the firewall to allow ALL UDP on AWS >>> >>> My SIP extension has >>> nat=force_rport,comedia >>> qualify=yes >>> allow=ulaw >>> allow=alaw >>> allow=gsm >>> canreinvite=yes >>> >>> I enable "rtp set debug on" and the console is printing info. >>> >>> The call comes into my linphone softphone - but I get no audio on my >>> linphone softphone. >>> What might I be missing to allow the audio ? >>> Volume is up. >>> >>> Thanks >>> >>> Jerry >>> >> >> >> I just noticed the RTP log is sending to 192.168.2.0 which is my local >> lan address of the linphone - it should be sending to the NAT address and >> is not. >> What did I not set correctly ? >> I am not using pjsip - but the older asterisk. >> >> Thanks >> >> Jerry >> > > >Have you configured chan_sip to know it is behind NAT itself and what its > >public IP address is? If not, then you'll get no audio. > > I'm thinking I have not. What did I miss ? > The sample configuration file outlines how things work, and the options for it: https://github.com/asterisk/asterisk/blob/master/configs/samples/sip.conf.sample#L874 in general localnet and externip (or externaddr, or externhost) -- Joshua C. Colp Asterisk Project Lead Sangoma Technologies Check us out at www.sangoma.com and www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Trying asterisk on AWS
On Thu, Oct 6, 2022 at 9:02 AM Jerry Geis wrote: > > > On Thu, Oct 6, 2022 at 8:59 AM Jerry Geis wrote: > >> I am trying to get audio to work on AWS using asterisk 18.14.0 >> >> I have enabled the firewall to allow ALL UDP on AWS >> >> My SIP extension has >> nat=force_rport,comedia >> qualify=yes >> allow=ulaw >> allow=alaw >> allow=gsm >> canreinvite=yes >> >> I enable "rtp set debug on" and the console is printing info. >> >> The call comes into my linphone softphone - but I get no audio on my >> linphone softphone. >> What might I be missing to allow the audio ? >> Volume is up. >> >> Thanks >> >> Jerry >> > > > I just noticed the RTP log is sending to 192.168.2.0 which is my local lan > address of the linphone - it should be sending to the NAT address and is > not. > What did I not set correctly ? > I am not using pjsip - but the older asterisk. > > Thanks > > Jerry > >Have you configured chan_sip to know it is behind NAT itself and what its >public IP address is? If not, then you'll get no audio. I'm thinking I have not. What did I miss ? Thanks, Jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Trying asterisk on AWS
On Thu, Oct 6, 2022 at 10:03 AM Jerry Geis wrote: > > > On Thu, Oct 6, 2022 at 8:59 AM Jerry Geis wrote: > >> I am trying to get audio to work on AWS using asterisk 18.14.0 >> >> I have enabled the firewall to allow ALL UDP on AWS >> >> My SIP extension has >> nat=force_rport,comedia >> qualify=yes >> allow=ulaw >> allow=alaw >> allow=gsm >> canreinvite=yes >> >> I enable "rtp set debug on" and the console is printing info. >> >> The call comes into my linphone softphone - but I get no audio on my >> linphone softphone. >> What might I be missing to allow the audio ? >> Volume is up. >> >> Thanks >> >> Jerry >> > > > I just noticed the RTP log is sending to 192.168.2.0 which is my local lan > address of the linphone - it should be sending to the NAT address and is > not. > What did I not set correctly ? > I am not using pjsip - but the older asterisk. > Have you configured chan_sip to know it is behind NAT itself and what its public IP address is? If not, then you'll get no audio. -- Joshua C. Colp Asterisk Project Lead Sangoma Technologies Check us out at www.sangoma.com and www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Trying asterisk on AWS
On Thu, Oct 6, 2022 at 8:59 AM Jerry Geis wrote: > I am trying to get audio to work on AWS using asterisk 18.14.0 > > I have enabled the firewall to allow ALL UDP on AWS > > My SIP extension has > nat=force_rport,comedia > qualify=yes > allow=ulaw > allow=alaw > allow=gsm > canreinvite=yes > > I enable "rtp set debug on" and the console is printing info. > > The call comes into my linphone softphone - but I get no audio on my > linphone softphone. > What might I be missing to allow the audio ? > Volume is up. > > Thanks > > Jerry > I just noticed the RTP log is sending to 192.168.2.0 which is my local lan address of the linphone - it should be sending to the NAT address and is not. What did I not set correctly ? I am not using pjsip - but the older asterisk. Thanks Jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Trying asterisk on AWS
I am trying to get audio to work on AWS using asterisk 18.14.0 I have enabled the firewall to allow ALL UDP on AWS My SIP extension has nat=force_rport,comedia qualify=yes allow=ulaw allow=alaw allow=gsm canreinvite=yes I enable "rtp set debug on" and the console is printing info. The call comes into my linphone softphone - but I get no audio on my linphone softphone. What might I be missing to allow the audio ? Volume is up. Thanks Jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk 8.14.0 and multicast sometimes not hear anythign
I am just doing a basic call in. exten => 140,1,Answer exten => 140,n,Playback(beep) exten => 140,n,Dial(MulticastRTP/basic/239.168.4.90:3040//t(15)) exten => 140,n,Hangup this works - but "sometimes" I get reports that "nothing" was heard. Is there anything special to do for multicast ? Any thoughts on why once in a great while nothing would be heard ? Thanks Jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
