[asterisk-users] app_jack.c - error - failing to write to ringbuffer
Hi all, I have been successufully using JACK application within Asterisk. Yet I am hearing quite some glitches in both input and output ends I am using jack with an external sound card RME HAmmerfall DSP Mutliface , but also tried the internal sound card, without a significant change. What seems to have improved a bit the quality was to change the jack settings to larger periods to its maximum 4096. There no longer sudden burst of noise, but the audio still choppy and distorted. And Asterisk console prints the following message: WARNING[1837][C-0003]: app_jack.c:598 queue_voice_frame: Tried to write 3532 bytes to the ringbuffer, but only wrote 2279 where the "only wrote" value changes between 0, 2279, 2255 jack is running with the following setting: jackd -d alsa -r 44100 -p 4096 -nperiod 3 -C hw:DSP -P hw:DSP -S I am on Asterisk 14.4.0 on Debian Jessie (8.7). The extension I am using: exten => 1234,1,Answer() same = n,JACK(i(system:playback_1),o(mpv:out_0)); Any ideas of what can be done to improve the audio quality when using JACK app? Thanks a -- oo.io bibliotecha.info -- _ -- 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] asterisk server - no sound
I am using version: 14.5.0 No, Im not using Dundi. Can you a bit more informative when you say I "need to configure the IPs in your server"? thanks! a On 06/06/2017 07:47 PM, Marcelo Terres wrote: > I think you need to configure the IPs in your server. You just have > localhost... > Marcelo H. Terres <mhter...@gmail.com> > IM: mhter...@jabber.mundoopensource.com.br > https://www.mundoopensource.com.br > https://twitter.com/mhterres > https://linkedin.com/in/marceloterres > > > On 6 June 2017 at 16:27, andre castro <an...@andrecastro.info> wrote: >> Thanks Anthony. >> >> I did it on the server, according to >> https://www.voip-info.org/wiki/view/port+forwarding >> >> However after doing it, when running Asterisk I get the following message >> sudo asterisk -vvr >> No ethernet interface found for seeding global EID. You will have to set >> it manually. >> Unable to access the running directory (No such file or directory). >> Changing to '/' for compatibility. >> >> How and where can it be set? >> >> My server ifconfig: >> >> loLink encap:Local Loopback >> inet addr:127.0.0.1 Mask:255.0.0.0 >> inet6 addr: ::1/128 Scope:Host >> UP LOOPBACK RUNNING MTU:65536 Metric:1 >> RX packets:113895058 errors:0 dropped:0 overruns:0 frame:0 >> TX packets:113895058 errors:0 dropped:0 overruns:0 carrier:0 >> collisions:0 txqueuelen:0 >> RX bytes:36041459269 (33.5 GiB) TX bytes:36041459269 (33.5 GiB) >> >> venet0Link encap:UNSPEC HWaddr >> 00-00-00-00-00-00-00-00-00-00-00-00-00-00-00-00 >> inet addr:127.0.0.1 P-t-P:127.0.0.1 Bcast:0.0.0.0 >> Mask:255.255.255.255 >> inet6 addr: ::2/128 Scope:Compat >> inet6 addr: 2a01:488:66:1000:5c33:846e:0:1/128 Scope:Global >> UP BROADCAST POINTOPOINT RUNNING NOARP MTU:1500 Metric:1 >> RX packets:158483849 errors:0 dropped:0 overruns:0 frame:0 >> TX packets:272193853 errors:0 dropped:230 overruns:0 carrier:0 >> collisions:0 txqueuelen:0 >> RX bytes:61233254724 (57.0 GiB) TX bytes:106403959440 (99.0 GiB) >> >> venet0:0 Link encap:UNSPEC HWaddr >> 00-00-00-00-00-00-00-00-00-00-00-00-00-00-00-00 >> inet addr:server.ip.add.r P-t-P:server.ip.add.r >> Bcast:server.ip.add.r Mask:255.255.255.255 >> UP BROADCAST POINTOPOINT RUNNING NOARP MTU:1500 Metric:1 >> >> >> >> On 06/06/2017 05:09 PM, Antony Stone wrote: >>> On Tuesday 06 June 2017 16:57:07 andre castro wrote: >>> >>>> On 06/06/2017 04:36 PM, Antony Stone wrote: >>>>> >>>>> Tell us about your networking arrangement - are both phones and the >>>>> Asterisk machine on the same network? >>>> >>>> Nop. They are in 2 different networks. The phones in one and the >>>> Asterisk machine in another. >>> >>> Okay, that is why you have audio between the two phones, then - they can see >>> each other directly, on the same network, and nothing is interfering with >>> the >>> traffic between them. >>> >>>>> Is there a router in between any of them? >>>> >>>> Yes. In the phones network. >>>> >>>>> Is there any NAT involved? >>>> >>>> Yes in the phones' network. They both have different private IP address >>>> and one public IP. >>> >>> Okay, I suspect that this NATting router is not passing the UDP packets from >>> the server back to the phones correctly, based on the SIP connection (when >>> the >>> phone makes the call). >>> >>> SIP is on UDP 5060; audio is on UDP 10,000 - 20,000. >>> >>> If it's a Linux router, you need to make sure you are allowing FORWARDed >>> traffic >>> which matches ESTABLISHED, RELATED. >>> >>> If it's not a Linux router, you need to find out how to get it to support >>> SIP >>> and RTSP. >>> >>> >>> Good luck, >>> >>> >>> Antony. >>> >> >> -- >> oo.io >> bibliotecha.info >> >> -- >> _ >> -- 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 > -- oo.io bibliotecha.info -- _ -- 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] asterisk server - no sound
Any ideas. After configuring port forwarding on the server (machine making nat) to forward connections originated from external clients to the machine running asterisk, as explained in https://www.voip-info.org/wiki/view/port+forwarding My peers were unable to register. And When running Asterisk I am getting: No ethernet interface found for seeding global EID. You will have to set it manually. Unable to access the running directory (No such file or directory). Changing to '/' for compatibility. Any advice what to do next? thanks a On 06/06/2017 05:27 PM, andre castro wrote: > Thanks Anthony. > > I did it on the server, according to > https://www.voip-info.org/wiki/view/port+forwarding > > However after doing it, when running Asterisk I get the following message > sudo asterisk -vvr > No ethernet interface found for seeding global EID. You will have to set > it manually. > Unable to access the running directory (No such file or directory). > Changing to '/' for compatibility. > > How and where can it be set? > > My server ifconfig: > > loLink encap:Local Loopback > inet addr:127.0.0.1 Mask:255.0.0.0 > inet6 addr: ::1/128 Scope:Host > UP LOOPBACK RUNNING MTU:65536 Metric:1 > RX packets:113895058 errors:0 dropped:0 overruns:0 frame:0 > TX packets:113895058 errors:0 dropped:0 overruns:0 carrier:0 > collisions:0 txqueuelen:0 > RX bytes:36041459269 (33.5 GiB) TX bytes:36041459269 (33.5 GiB) > > venet0Link encap:UNSPEC HWaddr > 00-00-00-00-00-00-00-00-00-00-00-00-00-00-00-00 > inet addr:127.0.0.1 P-t-P:127.0.0.1 Bcast:0.0.0.0 > Mask:255.255.255.255 > inet6 addr: ::2/128 Scope:Compat > inet6 addr: 2a01:488:66:1000:5c33:846e:0:1/128 Scope:Global > UP BROADCAST POINTOPOINT RUNNING NOARP MTU:1500 Metric:1 > RX packets:158483849 errors:0 dropped:0 overruns:0 frame:0 > TX packets:272193853 errors:0 dropped:230 overruns:0 carrier:0 > collisions:0 txqueuelen:0 > RX bytes:61233254724 (57.0 GiB) TX bytes:106403959440 (99.0 GiB) > > venet0:0 Link encap:UNSPEC HWaddr > 00-00-00-00-00-00-00-00-00-00-00-00-00-00-00-00 > inet addr:server.ip.add.r P-t-P:server.ip.add.r > Bcast:server.ip.add.r Mask:255.255.255.255 > UP BROADCAST POINTOPOINT RUNNING NOARP MTU:1500 Metric:1 > > > > On 06/06/2017 05:09 PM, Antony Stone wrote: >> On Tuesday 06 June 2017 16:57:07 andre castro wrote: >> >>> On 06/06/2017 04:36 PM, Antony Stone wrote: >>>> >>>> Tell us about your networking arrangement - are both phones and the >>>> Asterisk machine on the same network? >>> >>> Nop. They are in 2 different networks. The phones in one and the >>> Asterisk machine in another. >> >> Okay, that is why you have audio between the two phones, then - they can see >> each other directly, on the same network, and nothing is interfering with >> the >> traffic between them. >> >>>> Is there a router in between any of them? >>> >>> Yes. In the phones network. >>> >>>> Is there any NAT involved? >>> >>> Yes in the phones' network. They both have different private IP address >>> and one public IP. >> >> Okay, I suspect that this NATting router is not passing the UDP packets from >> the server back to the phones correctly, based on the SIP connection (when >> the >> phone makes the call). >> >> SIP is on UDP 5060; audio is on UDP 10,000 - 20,000. >> >> If it's a Linux router, you need to make sure you are allowing FORWARDed >> traffic >> which matches ESTABLISHED, RELATED. >> >> If it's not a Linux router, you need to find out how to get it to support >> SIP >> and RTSP. >> >> >> Good luck, >> >> >> Antony. >> > -- oo.io bibliotecha.info -- _ -- 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] asterisk server - no sound
Thanks Anthony. I did it on the server, according to https://www.voip-info.org/wiki/view/port+forwarding However after doing it, when running Asterisk I get the following message sudo asterisk -vvr No ethernet interface found for seeding global EID. You will have to set it manually. Unable to access the running directory (No such file or directory). Changing to '/' for compatibility. How and where can it be set? My server ifconfig: loLink encap:Local Loopback inet addr:127.0.0.1 Mask:255.0.0.0 inet6 addr: ::1/128 Scope:Host UP LOOPBACK RUNNING MTU:65536 Metric:1 RX packets:113895058 errors:0 dropped:0 overruns:0 frame:0 TX packets:113895058 errors:0 dropped:0 overruns:0 carrier:0 collisions:0 txqueuelen:0 RX bytes:36041459269 (33.5 GiB) TX bytes:36041459269 (33.5 GiB) venet0Link encap:UNSPEC HWaddr 00-00-00-00-00-00-00-00-00-00-00-00-00-00-00-00 inet addr:127.0.0.1 P-t-P:127.0.0.1 Bcast:0.0.0.0 Mask:255.255.255.255 inet6 addr: ::2/128 Scope:Compat inet6 addr: 2a01:488:66:1000:5c33:846e:0:1/128 Scope:Global UP BROADCAST POINTOPOINT RUNNING NOARP MTU:1500 Metric:1 RX packets:158483849 errors:0 dropped:0 overruns:0 frame:0 TX packets:272193853 errors:0 dropped:230 overruns:0 carrier:0 collisions:0 txqueuelen:0 RX bytes:61233254724 (57.0 GiB) TX bytes:106403959440 (99.0 GiB) venet0:0 Link encap:UNSPEC HWaddr 00-00-00-00-00-00-00-00-00-00-00-00-00-00-00-00 inet addr:server.ip.add.r P-t-P:server.ip.add.r Bcast:server.ip.add.r Mask:255.255.255.255 UP BROADCAST POINTOPOINT RUNNING NOARP MTU:1500 Metric:1 On 06/06/2017 05:09 PM, Antony Stone wrote: > On Tuesday 06 June 2017 16:57:07 andre castro wrote: > >> On 06/06/2017 04:36 PM, Antony Stone wrote: >>> >>> Tell us about your networking arrangement - are both phones and the >>> Asterisk machine on the same network? >> >> Nop. They are in 2 different networks. The phones in one and the >> Asterisk machine in another. > > Okay, that is why you have audio between the two phones, then - they can see > each other directly, on the same network, and nothing is interfering with the > traffic between them. > >>> Is there a router in between any of them? >> >> Yes. In the phones network. >> >>> Is there any NAT involved? >> >> Yes in the phones' network. They both have different private IP address >> and one public IP. > > Okay, I suspect that this NATting router is not passing the UDP packets from > the server back to the phones correctly, based on the SIP connection (when > the > phone makes the call). > > SIP is on UDP 5060; audio is on UDP 10,000 - 20,000. > > If it's a Linux router, you need to make sure you are allowing FORWARDed > traffic > which matches ESTABLISHED, RELATED. > > If it's not a Linux router, you need to find out how to get it to support SIP > and RTSP. > > > Good luck, > > > Antony. > -- oo.io bibliotecha.info -- _ -- 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] asterisk server - no sound
Thank you Daniel for pointing out the errors and debug option. Both fixed and on. It made no difference. There are no errors printed and still no sound on ppers Now to Antony questions: On 06/06/2017 04:36 PM, Antony Stone wrote: > On Tuesday 06 June 2017 15:18:32 andre castro wrote: > >> I just installed asterisk in a debian server. >> All seems to be running fine, but the audio sent by the server. > >> But I hear nothing at the peer's end. >> >> When one peer calls another, sound comes through just fine. > > Tell us about your networking arrangement - are both phones and the Asterisk > machine on the same network? Nop. They are in 2 different networks. The phones in one and the Asterisk machine in another. > > Is there a router in between any of them? Yes. In the phones network. > > Is there any NAT involved? Yes in the phones' network. They both have different private IP address and one public IP. > >> Do I need to have alsa installed?? > > No. So I thought. Thanks guys!! > > > Antony. > -- oo.io bibliotecha.info -- _ -- 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 server - no sound
hello folks, this might be a simple question... I just installed asterisk in a debian server. All seems to be running fine, but the audio sent by the server. If I have one of my registered peers call and extension (102) that plays back audio (extension.conf and sip.conf coffee-pasted below), Asterisk answers and prints no errors. Its `sip show channels` prints: PeerUser/ANRCall IDFormatHoldLast MessageExpiry Peer peer.ip1001 1...-5060 (ulaw) No Rx: ACK 1001 But I hear nothing at the peer's end. When one peer calls another, sound comes through just fine. So my hunch is that is something to do with the audio supplied by the server. Do I need to have alsa installed?? Any hint? sip.conf: [general] context = unauthenticated bindport = 5060 bindaddr = 0.0.0.0 tcpbindaddr = 0.0.0.0 tcpenable = yes videosupport = no textsupport=yes alwaysauthreject=yes allowguest=no [1001] ; grandstream 1 context = home type = friend callerid = One <1001> secret = XYZ host = dynamic mailbox = 1001 disallow = all allow = ulaw transport = udp dtmfmode=auto ; accept touch-tones from the devices, negotiated automatically nat=force_rport [1005] ; mobile context = home type = friend callerid = Five <1005> secret = XYZ host = dynamic mailbox = 1005 disallow = all allow = ulaw transport = udp dtmfmode=auto ; accept touch-tones from the devices, negotiated automatically nat=force_rport extensions.conf: [home] exten = 102,1,Answer() same = n,Wait(1) same = n,Playback(beep) same = n,Wait(1) same = n,Playback(im-sorry) same = n,Wait(1) same = n,Playback(number-not-answering) same = n,Wait(1) same = n,Hangup() exten => 1001,1,Dial(SIP/1001) ; grandstream phone exten => 1005,1,Dial(SIP/1005) ; mobile -- oo.io bibliotecha.info -- _ -- 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] JACK_HOOK Auto fallthrough
Hi,
I am having problems adding a JACK_HOOK function to a Asterisk dialplan.
I have jackd (1.9.10) running. And tested.
in extensions.conf I have the following plan:
exten = 99,1,Answer()
same = n, Set(JACK_HOOK(manipulate,s(default),i(system:playback_1))=on)
When I set a peer to call it (from Keriga), the Asterisk console, prints:
-- Executing [99@home:1] Answer("SIP/1002-0033", "") in new stack
-- Executing [99@home:2] Set("SIP/1002-0033",
"JACK_HOOK(manipulate,s(default),i(system:playback_1))=on") in new stack
[May 11 21:55:26] NOTICE[6526][C-004e]: app_jack.c:192
log_jack_status: Client Open Status: Failure, Server Failed
-- Auto fallthrough, channel 'SIP/1002-0033' status is 'UNKNOWN'
I don't understand why does "Client Open Status: Failure, Server Failed"
appears.
And also 'SIP/1002-0033' status is 'UNKNOWN'.
When checking jack_lsp I can only see the system ports, but no Asterisk
port.
Any hints on how I could solve this are appreciated.
Thanks
a
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Re: [asterisk-users] app_jack unavailable
Thanks Joshua, rerun ./configure did the job. On 05/10/2017 02:48 PM, Joshua Colp wrote: > On Wed, May 10, 2017, at 09:33 AM, andre castro wrote: >> Thanks J. >> It didn't work. >> >> On 05/10/2017 01:57 PM, J Montoya or A J Stiles wrote: >>> On Wednesday 10 May 2017, andre castro wrote: >>>> Hello, >>>> I am new to Asterisk, so please bear with me. >>>> I have made a success installation from source of Asterisk 14.4.0 on >>>> Debian Jessie (8.7). And I am running the Asterisk server, with several >>>> extensions and dialplans, all working well. >>>> >>>> However I am struggling to get app_jack to run. >>>> >>>> In menuselect I can see that it is XXX due to dependencies on jack and >>>> resample, however both Debian packages: resample and jackd are installed >>>> in this machine. >>>> >>>> Is there something that I am missing? >>> >>> Most probably some -dev files. (Come on, distro folks, it's 2017; >>> bandwidth >>> and storage are way cheaper than they were, back when the decision was >>> first >>> made to separate out files only needed by developers into their own >>> packages, >>> and now it's causing more grief than it is saving.) >> >> Indeed. apt-get install libjack-dev libresample-dev were not installed. >> libjack-dev libresample-dev , so I installed. >> In the installation of libresample-dev apt-get selected >> 'libresample1-dev' instead of 'libresample-dev'. Not sure if that is a >> problem. >> Also installed libjack-jackd2-dev. >> >> But when I run menuselect in Applications, app_jack still is XXX >> mentioning jack and resample unmet dependencies. >> >> Is there any extra step i need to do? > > You need to rerun ./configure so that it is aware of what you have > installed. > -- oo.io bibliotecha.info -- _ -- 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] app_jack unavailable
Thanks J. It didn't work. On 05/10/2017 01:57 PM, J Montoya or A J Stiles wrote: > On Wednesday 10 May 2017, andre castro wrote: >> Hello, >> I am new to Asterisk, so please bear with me. >> I have made a success installation from source of Asterisk 14.4.0 on >> Debian Jessie (8.7). And I am running the Asterisk server, with several >> extensions and dialplans, all working well. >> >> However I am struggling to get app_jack to run. >> >> In menuselect I can see that it is XXX due to dependencies on jack and >> resample, however both Debian packages: resample and jackd are installed >> in this machine. >> >> Is there something that I am missing? > > Most probably some -dev files. (Come on, distro folks, it's 2017; bandwidth > and storage are way cheaper than they were, back when the decision was first > made to separate out files only needed by developers into their own packages, > and now it's causing more grief than it is saving.) Indeed. apt-get install libjack-dev libresample-dev were not installed. libjack-dev libresample-dev , so I installed. In the installation of libresample-dev apt-get selected 'libresample1-dev' instead of 'libresample-dev'. Not sure if that is a problem. Also installed libjack-jackd2-dev. But when I run menuselect in Applications, app_jack still is XXX mentioning jack and resample unmet dependencies. Is there any extra step i need to do? a > >> What do I need to do in order to have the jack module to run? > > Probably just > > # apt-get install libjack-dev libresample-dev > > although it's possible that you might need some additional -dev packages. > But > try the above first. > > -- oo.io bibliotecha.info -- _ -- 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] app_jack unavailable
Hello, I am new to Asterisk, so please bear with me. I have made a success installation from source of Asterisk 14.4.0 on Debian Jessie (8.7). And I am running the Asterisk server, with several extensions and dialplans, all working well. However I am struggling to get app_jack to run. In menuselect I can see that it is XXX due to dependencies on jack and resample, however both Debian packages: resample and jackd are installed in this machine. Is there something that I am missing? What do I need to do in order to have the jack module to run? Cheers -- oo.io bibliotecha.info -- _ -- 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
