Re: [asterisk-users] Running on virtual machine and audio not intelligable
why stick with version 11 ? upgrade to 13 is a starting solution. On Jan 25, 2018 3:28 AM, "Richard Mudgett"wrote: > > > On Wed, Jan 24, 2018 at 7:55 PM, Jerry Geis wrote: > >> >why load or even install dahdi if no cards are used? >> >> I thought dahdi was needed for a timing source. Doesn't ConfBridge need a >> timing source? >> > > DAHDI is required for MeetMe to do the audio mixing. ConfBridge does not > need DAHDI since > it does its own mixing in Asterisk. > > As for timing sources you have several to choose from of which DAHDI is > one of them. See > menuselect res_timing_xxx modules. Timing is really only needed when > playing back sound > prompts when nothing is being received. > > Richard > > > -- > _ > -- 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 > -- _ -- 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] Running on virtual machine and audio not intelligable
On Wed, Jan 24, 2018 at 7:55 PM, Jerry Geiswrote: > >why load or even install dahdi if no cards are used? > > I thought dahdi was needed for a timing source. Doesn't ConfBridge need a > timing source? > DAHDI is required for MeetMe to do the audio mixing. ConfBridge does not need DAHDI since it does its own mixing in Asterisk. As for timing sources you have several to choose from of which DAHDI is one of them. See menuselect res_timing_xxx modules. Timing is really only needed when playing back sound prompts when nothing is being received. Richard -- _ -- 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] Running on virtual machine and audio not intelligable
>why load or even install dahdi if no cards are used? I thought dahdi was needed for a timing source. Doesn't ConfBridge need a timing source? 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] Running on virtual machine and audio not intelligable
why load or even install dahdi if no cards are used? On Jan 24, 2018 10:30 PM, "Jerry Geis"wrote: > Hi All, > Running asterisk 11.25.3, the /proc/cpuinfo says > Intel(R) Xeon(R) CPU X5675 @ 3.07GHz > > lsmod | grep dahdi gives > dahdi_transcode16384 1 wctc4xxp > dahdi_voicebus 61440 2 wctdm24xxp,wcte12xp > dahdi 225280 11 > wctdm24xxp,wcfxo,wctdm,dahdi_transcode,oct612x > dahdi_voicebus,wcb4xxp,wct1xxp,wct4xxp,wcte11xp,wcte12xp > crc_ccitt 16384 2 wctdm24xxp,dahdi > > There are no cards in the VM - just the CentOS 7.4 operating system. I'm > actually running 4.4.92 from elrepo as the kernel. > > I am using the confbridge with about 100 devices in the "one way" > conference and the audio is not intelligible. Sending audio to a single > endpoint is fine. > > dahdi_test running when nothing is happening is 99.8 running with the ALL > 100 units in conf is about 99.6. > > What can I do do get audio where it needs to be ? I believe the > environment is VMware. > > 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 > -- _ -- 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] Running on virtual machine and audio not intelligable
Hi All, Running asterisk 11.25.3, the /proc/cpuinfo says Intel(R) Xeon(R) CPU X5675 @ 3.07GHz lsmod | grep dahdi gives dahdi_transcode16384 1 wctc4xxp dahdi_voicebus 61440 2 wctdm24xxp,wcte12xp dahdi 225280 11 wctdm24xxp,wcfxo,wctdm,dahdi_transcode,oct612x dahdi_voicebus,wcb4xxp,wct1xxp,wct4xxp,wcte11xp,wcte12xp crc_ccitt 16384 2 wctdm24xxp,dahdi There are no cards in the VM - just the CentOS 7.4 operating system. I'm actually running 4.4.92 from elrepo as the kernel. I am using the confbridge with about 100 devices in the "one way" conference and the audio is not intelligible. Sending audio to a single endpoint is fine. dahdi_test running when nothing is happening is 99.8 running with the ALL 100 units in conf is about 99.6. What can I do do get audio where it needs to be ? I believe the environment is VMware. 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] res_pjsip_transport_management.c: Shutting down transport
On Wed, Jan 24, 2018 at 7:07 AM, marek cervenkawrote: > hello, > > i met with this interesting situation > > [Jan 24 13:48:37] NOTICE[1049] res_pjsip_transport_management.c: Shutting > down transport '8' since no request was received in 32 seconds > > [Jan 24 13:48:37] NOTICE[1049] res_pjsip_transport_management.c: Shutting > down transport '8' since no request was received in 32 seconds > [Jan 24 13:48:41] NOTICE[1049] res_pjsip_transport_management.c: Shutting > down transport '' since no request was received in 32 seconds > [Jan 24 13:48:44] NOTICE[1049] res_pjsip_transport_management.c: Shutting > down transport 'e=" 79-ad2e-c47e6a3db178>";expires=60 > u▒l^' since no request was received in 32 seconds > [Jan 24 13:48:44] NOTICE[1049] res_pjsip_transport_management.c: Shutting > down transport '' since no request was received in 32 seconds > [Jan 24 13:48:45] NOTICE[1049] res_pjsip_transport_management.c: Shutting > down transport '' since no request was received in 32 seconds > [Jan 24 13:48:47] NOTICE[1049] res_pjsip_transport_management.c: Shutting > down transport 'e=" 37-966d-9a936a350728>";expires=60 > ' since no request was received in 32 seconds > [Jan 24 13:48:49] NOTICE[1049] res_pjsip_transport_management.c: Shutting > down transport '' since no request was received in 32 seconds > [Jan 24 13:48:50] NOTICE[1049] res_pjsip_transport_management.c: Shutting > down transport '.0 > Date: Wed, 24 Jan 2018 12:48:18 GMT > Allow: INVITE, ACK, CAN' since no request was received in 32 seconds > [Jan 24 13:48:53] NOTICE[1049] res_pjsip_transport_management.c: Shutting > down transport '' since no request was received in 32 seconds > [Jan 24 13:48:53] NOTICE[1049] res_pjsip_transport_management.c: Shutting > down transport ' SUBSCRIBE, INFO' since no request was received in 32 > seconds > [Jan 24 13:48:53] NOTICE[1049] res_pjsip_transport_management.c: Shutting > down transport 'c732305e-f905-489a-a6f4-5164f0809c8a>";expires=60 > Expires: 60 > @u▒^' since no request was received in 32 seconds > [Jan 24 13:49:27] NOTICE[1049] res_pjsip_transport_management.c: Shutting > down transport '▒▒<%▒▒*W▒▒▒$@▒▒▒{▒X_DL▒▒▒1▒▒"▒`$▒zC▒l▒o▒O▒3▒▒c:133 > idle_sched_cb: Shutting down transport '=" b-b2ca-6292f151c7c2>";expires=60 > > asterisk went crazy and had to be restarted > That module does 2 things. First it handles the keepalives if keep_alive_interval is > 0 in the pjsip.conf/global. It also attempts to mitigate DOS attacks if an attacker floods asterisk with TCP (or TLS) connections but doesn't send any actual messages within the time set in pjsip.conf/system/timer_b. When a connection is opened, a timer is started and if there is no recognizable SIP message before the timer expires, you get the "Shutting down transport" message. > > topology > > asterisk 13.18.2 + pjsip realtime + mariadb (mariadb is on different > network!) + jssip via wss as client > > extconfig.conf > > ps_endpoints => odbc,configDb > ps_auths => odbc,configDb > ps_aors => odbc,configDb > ps_domain_aliases => odbc,configDb > > sorcery.conf > > [res_pjsip] ; Realtime PJSIP configuration wizard > endpoint/cache=memory_cache,expire_on_reload=yes,full_backend_cache=yes > endpoint=realtime,ps_endpoints > auth/cache=memory_cache,expire_on_reload=yes,full_backend_cache=yes > auth=realtime,ps_auths > aor/cache=memory_cache,expire_on_reload=yes,full_backend_cache=yes > aor=realtime,ps_aors > domain_alias=realtime,ps_domain_aliases > > > there was net interruption on ~13:48 > > do you have any ideas what can be cause of "res_pjsip_transport_management.c: > Shutting down transport" ? > Yep, it was probably that network interruption. The incoming messages were being corrupted and not recognized as real SIP messages so the timer expired and the transports were shut down. > > my idea was that Asterisk with cache doesnt need realtime connectivity > with mariadb (can survive short internet interruptions) > > Marek > > > > > -- > _ > -- 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 -- George Joseph Digium, Inc. | Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & 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:
[asterisk-users] res_pjsip_transport_management.c: Shutting down transport
hello, i met with this interesting situation [Jan 24 13:48:37] NOTICE[1049] res_pjsip_transport_management.c: Shutting down transport '8' since no request was received in 32 seconds [Jan 24 13:48:37] NOTICE[1049] res_pjsip_transport_management.c: Shutting down transport '8' since no request was received in 32 seconds [Jan 24 13:48:41] NOTICE[1049] res_pjsip_transport_management.c: Shutting down transport '' since no request was received in 32 seconds [Jan 24 13:48:44] NOTICE[1049] res_pjsip_transport_management.c: Shutting down transport 'e="";expires=60 u▒l^' since no request was received in 32 seconds [Jan 24 13:48:44] NOTICE[1049] res_pjsip_transport_management.c: Shutting down transport '' since no request was received in 32 seconds [Jan 24 13:48:45] NOTICE[1049] res_pjsip_transport_management.c: Shutting down transport '' since no request was received in 32 seconds [Jan 24 13:48:47] NOTICE[1049] res_pjsip_transport_management.c: Shutting down transport 'e="";expires=60 ' since no request was received in 32 seconds [Jan 24 13:48:49] NOTICE[1049] res_pjsip_transport_management.c: Shutting down transport '' since no request was received in 32 seconds [Jan 24 13:48:50] NOTICE[1049] res_pjsip_transport_management.c: Shutting down transport '.0 Date: Wed, 24 Jan 2018 12:48:18 GMT Allow: INVITE, ACK, CAN' since no request was received in 32 seconds [Jan 24 13:48:53] NOTICE[1049] res_pjsip_transport_management.c: Shutting down transport '' since no request was received in 32 seconds [Jan 24 13:48:53] NOTICE[1049] res_pjsip_transport_management.c: Shutting down transport ' SUBSCRIBE, INFO' since no request was received in 32 seconds [Jan 24 13:48:53] NOTICE[1049] res_pjsip_transport_management.c: Shutting down transport 'c732305e-f905-489a-a6f4-5164f0809c8a>";expires=60 Expires: 60 @u▒^' since no request was received in 32 seconds [Jan 24 13:49:27] NOTICE[1049] res_pjsip_transport_management.c: Shutting down transport '▒▒<%▒▒*W▒▒▒$@▒▒▒{▒X_DL▒▒▒1▒▒"▒`$▒zC▒l▒o▒O▒3▒▒c:133 idle_sched_cb: Shutting down transport '="";expires=60 asterisk went crazy and had to be restarted topology asterisk 13.18.2 + pjsip realtime + mariadb (mariadb is on different network!) + jssip via wss as client extconfig.conf ps_endpoints => odbc,configDb ps_auths => odbc,configDb ps_aors => odbc,configDb ps_domain_aliases => odbc,configDb sorcery.conf [res_pjsip] ; Realtime PJSIP configuration wizard endpoint/cache=memory_cache,expire_on_reload=yes,full_backend_cache=yes endpoint=realtime,ps_endpoints auth/cache=memory_cache,expire_on_reload=yes,full_backend_cache=yes auth=realtime,ps_auths aor/cache=memory_cache,expire_on_reload=yes,full_backend_cache=yes aor=realtime,ps_aors domain_alias=realtime,ps_domain_aliases there was net interruption on ~13:48 do you have any ideas what can be cause of "res_pjsip_transport_management.c: Shutting down transport" ? my idea was that Asterisk with cache doesnt need realtime connectivity with mariadb (can survive short internet interruptions) Marek -- _ -- 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