Re: [asterisk-users] Odd issue CentOS 7 vs Ubuntu 20.04
On Fri, Feb 4, 2022 at 12:42 PM Jerry Geis wrote: > > > On Wed, Feb 2, 2022 at 1:06 PM Jerry Geis wrote: > >> >> >> On Wed, Feb 2, 2022 at 10:44 AM Jerry Geis wrote: >> >>> >>> >>> On Wed, Feb 2, 2022 at 9:26 AM Jerry Geis wrote: >>> So I have CentOS 7 server running asterisk 18.8.0 - all is good. I unplug that server - plug in a ubuntu 20.04 server at the same IP address. let my 3 devices reconnect to the ubuntu server When I pick up the polycom phone and dial it connects. I hear the other ends 'tone" - but when I press digits - nothing happens (to select a port) Seems everything is set for rfc2833. The devices are a TOA SIP Gateway, and a TOA N-8000 device connected to the GW. I have compared the settings of the polycom extension on both boxes - they match and also the SIP gateway. I tried to compare the sip debug from the Ubuntu to the centos and "looked" the same to me. Where might I look next or what might I look at ? Thanks, Jerry >>> >>> >>> ok - if I "rtp set debug on " on the CentOS 7 server I get a tone of >>> logging. >>> >>> if I do the same on the ubuntu 20.04 all i get is like 2 lines. >>> I have done "systemctl stop firewalld" on the ubuntu box - same result. >>> >>> Where do I look next ? >>> >>> Jerry >>> >> >> >> I dont get it - I certainly getting RTP traffic because I defined an >> extension to playback the demo-congrats messages. >> I call that extension - and ALL kinds of RTP traffic prints on teh >> console. >> >> But when I call the one extension - 103 - all it prints is 2 lines. >> >> I also removed the source tree - un tarred - ran the >> contrib/scripts/install_prereq install script, it did install a couple >> packages - I dont think they mattered. >> do the ./configure, make, make install and started up again - same issue >> though. >> >> Jerry >> > > > > So - still on this... > > I was just dialing the SIP Gateway with Dial(SIP/103) > > if I change my Dial command to this: > > Dial(SIP/103,20,D(15)) > So I send out the DTMF in the dial command - this works and connects me > and the DTMF is delivered and I get the right port. > > The problem still remains - Dialing just Dial(SIP/103) from the polycom > phone - and then doing 15 for DTMF does not work. Cant figure out why ? > > Any thoughts ? > > Jerry > This ended up being a simple canreinvite situation... I had yes - and needed to be set to NO. 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] Odd issue CentOS 7 vs Ubuntu 20.04
>The usage of D(15) causes Asterisk to produce RTP on its own. Without it, >it merely forwards RTP. If a NAT/firewall requires media to be sent before >allowing media in, then you'll have no media flow. You can use the >"rtpkeepalive" option to have the RTP stack produce keepalive packets, >which will then open the NAT/firewall. -- Hi Josh - Thanks - I have also turned off the firewall with "systemctl stop firewalld". Did not make a differernce. I am not at the site to change the to rtpkeepalive - will try that monday. 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] Odd issue CentOS 7 vs Ubuntu 20.04
On Fri, Feb 4, 2022 at 1:42 PM Jerry Geis wrote: > > So - still on this... > > I was just dialing the SIP Gateway with Dial(SIP/103) > > if I change my Dial command to this: > > Dial(SIP/103,20,D(15)) > So I send out the DTMF in the dial command - this works and connects me > and the DTMF is delivered and I get the right port. > > The problem still remains - Dialing just Dial(SIP/103) from the polycom > phone - and then doing 15 for DTMF does not work. Cant figure out why ? > > Any thoughts ? > The usage of D(15) causes Asterisk to produce RTP on its own. Without it, it merely forwards RTP. If a NAT/firewall requires media to be sent before allowing media in, then you'll have no media flow. You can use the "rtpkeepalive" option to have the RTP stack produce keepalive packets, which will then open the NAT/firewall. -- Joshua C. Colp Asterisk Technical 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] Odd issue CentOS 7 vs Ubuntu 20.04
On Wed, Feb 2, 2022 at 1:06 PM Jerry Geis wrote: > > > On Wed, Feb 2, 2022 at 10:44 AM Jerry Geis wrote: > >> >> >> On Wed, Feb 2, 2022 at 9:26 AM Jerry Geis wrote: >> >>> So I have CentOS 7 server running asterisk 18.8.0 - all is good. >>> >>> I unplug that server - plug in a ubuntu 20.04 server at the same IP >>> address. >>> let my 3 devices reconnect to the ubuntu server >>> >>> When I pick up the polycom phone and dial it connects. >>> I hear the other ends 'tone" - but when I press digits - nothing happens >>> (to select a port) >>> Seems everything is set for rfc2833. >>> >>> The devices are a TOA SIP Gateway, and a TOA N-8000 device connected to >>> the GW. >>> >>> I have compared the settings of the polycom extension on both boxes - >>> they match and also the SIP gateway. >>> >>> I tried to compare the sip debug from the Ubuntu to the centos and >>> "looked" the same to me. >>> >>> Where might I look next or what might I look at ? >>> >>> Thanks, >>> >>> Jerry >>> >> >> >> ok - if I "rtp set debug on " on the CentOS 7 server I get a tone of >> logging. >> >> if I do the same on the ubuntu 20.04 all i get is like 2 lines. >> I have done "systemctl stop firewalld" on the ubuntu box - same result. >> >> Where do I look next ? >> >> Jerry >> > > > I dont get it - I certainly getting RTP traffic because I defined an > extension to playback the demo-congrats messages. > I call that extension - and ALL kinds of RTP traffic prints on teh console. > > But when I call the one extension - 103 - all it prints is 2 lines. > > I also removed the source tree - un tarred - ran the > contrib/scripts/install_prereq install script, it did install a couple > packages - I dont think they mattered. > do the ./configure, make, make install and started up again - same issue > though. > > Jerry > So - still on this... I was just dialing the SIP Gateway with Dial(SIP/103) if I change my Dial command to this: Dial(SIP/103,20,D(15)) So I send out the DTMF in the dial command - this works and connects me and the DTMF is delivered and I get the right port. The problem still remains - Dialing just Dial(SIP/103) from the polycom phone - and then doing 15 for DTMF does not work. Cant figure out why ? Any thoughts ? 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] Odd issue CentOS 7 vs Ubuntu 20.04
On Wed, Feb 2, 2022 at 10:44 AM Jerry Geis wrote: > > > On Wed, Feb 2, 2022 at 9:26 AM Jerry Geis wrote: > >> So I have CentOS 7 server running asterisk 18.8.0 - all is good. >> >> I unplug that server - plug in a ubuntu 20.04 server at the same IP >> address. >> let my 3 devices reconnect to the ubuntu server >> >> When I pick up the polycom phone and dial it connects. >> I hear the other ends 'tone" - but when I press digits - nothing happens >> (to select a port) >> Seems everything is set for rfc2833. >> >> The devices are a TOA SIP Gateway, and a TOA N-8000 device connected to >> the GW. >> >> I have compared the settings of the polycom extension on both boxes - >> they match and also the SIP gateway. >> >> I tried to compare the sip debug from the Ubuntu to the centos and >> "looked" the same to me. >> >> Where might I look next or what might I look at ? >> >> Thanks, >> >> Jerry >> > > > ok - if I "rtp set debug on " on the CentOS 7 server I get a tone of > logging. > > if I do the same on the ubuntu 20.04 all i get is like 2 lines. > I have done "systemctl stop firewalld" on the ubuntu box - same result. > > Where do I look next ? > > Jerry > I dont get it - I certainly getting RTP traffic because I defined an extension to playback the demo-congrats messages. I call that extension - and ALL kinds of RTP traffic prints on teh console. But when I call the one extension - 103 - all it prints is 2 lines. I also removed the source tree - un tarred - ran the contrib/scripts/install_prereq install script, it did install a couple packages - I dont think they mattered. do the ./configure, make, make install and started up again - same issue though. 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] Odd issue CentOS 7 vs Ubuntu 20.04
On Wed, Feb 2, 2022 at 9:26 AM Jerry Geis wrote: > So I have CentOS 7 server running asterisk 18.8.0 - all is good. > > I unplug that server - plug in a ubuntu 20.04 server at the same IP > address. > let my 3 devices reconnect to the ubuntu server > > When I pick up the polycom phone and dial it connects. > I hear the other ends 'tone" - but when I press digits - nothing happens > (to select a port) > Seems everything is set for rfc2833. > > The devices are a TOA SIP Gateway, and a TOA N-8000 device connected to > the GW. > > I have compared the settings of the polycom extension on both boxes - they > match and also the SIP gateway. > > I tried to compare the sip debug from the Ubuntu to the centos and > "looked" the same to me. > > Where might I look next or what might I look at ? > > Thanks, > > Jerry > ok - if I "rtp set debug on " on the CentOS 7 server I get a tone of logging. if I do the same on the ubuntu 20.04 all i get is like 2 lines. I have done "systemctl stop firewalld" on the ubuntu box - same result. Where do I look next ? 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] Odd issue CentOS 7 vs Ubuntu 20.04
So I have CentOS 7 server running asterisk 18.8.0 - all is good. I unplug that server - plug in a ubuntu 20.04 server at the same IP address. let my 3 devices reconnect to the ubuntu server When I pick up the polycom phone and dial it connects. I hear the other ends 'tone" - but when I press digits - nothing happens (to select a port) Seems everything is set for rfc2833. The devices are a TOA SIP Gateway, and a TOA N-8000 device connected to the GW. I have compared the settings of the polycom extension on both boxes - they match and also the SIP gateway. I tried to compare the sip debug from the Ubuntu to the centos and "looked" the same to me. Where might I look next or what might I look at ? 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
