{"action":"created","issue":{"url":"https://api.github.com/repos/openwrt/telephony/issues/7","labels_url":"https://api.github.com/repos/openwrt/telephony/issues/7/labels{/name}","comments_url":"https://api.github.com/repos/openwrt/telephony/issues/7/comments","events_url":"https://api.github.com/repos/openwrt/telephony/issues/7/events","html_url":"https://github.com/openwrt/telephony/issues/7","id":53386324,"number":7,"title":"asterisk11-chan-dongle : broken audio channel Asterisk => GSM","user":{"login":"ljakob","id":4210026,"avatar_url":"https://avatars.githubusercontent.com/u/4210026?v=3","gravatar_id":"","url":"https://api.github.com/users/ljakob","html_url":"https://github.com/ljakob","followers_url":"https://api.github.com/users/ljakob/followers","following_url":"https://api.github.com/users/ljakob/following{/other_user}","gists_url":"https://api.github.com/users/ljakob/gists{/gist_id}","starred_url":"https://api.github.com/users/ljakob/starred{/owner}{/repo}","subscriptions_url":"https://api.github.com/users/ljakob/subscriptions","organizations_url":"https://api.github.com/users/ljakob/orgs","repos_url":"https://api.github.com/users/ljakob/repos","events_url":"https://api.github.com/users/ljakob/events{/privacy}","received_events_url":"https://api.github.com/users/ljakob/received_events","type":"User","site_admin":false},"labels":[],"state":"open","locked":false,"assignee":null,"milestone":null,"comments":13,"created_at":"2015-01-05T11:17:28Z","updated_at":"2015-03-07T10:42:46Z","closed_at":null,"body":"Recreated from: https://dev.openwrt.org/ticket/18672\r\nProblem is on: Barrier Breaker 14.07\r\n\r\nHi,\r\n\r\nI've moved my asterisk installation (Debian x64) to my OpenWrt-Box:\r\n\r\n```\r\nsystem type             : Qualcomm Atheros QCA9558 rev 0\r\nmachine                 : TP-LINK Archer C7\r\ncpu model               : MIPS 74Kc V5.0\r\n```\r\n\r\nAll is working fine, but the chan_dongle has a broken audio channel for audio from asterisk to the GSM network, the other direction is fine.\r\n\r\nHardware is:\r\n```\r\nXXX*CLI> dongle show devices\r\nID           Group State      RSSI Mode Submode Provider Name  Model      Firmware          IMEI             IMSI             Number\r\nXXX   0     Free       19   0    0       Orange CH@     K3715      11.104.16.81.00   353435027169XXX  228031700051XXX  Unknown\r\n\r\n```\r\n\r\nA simple test is:\r\n\r\nSetup chan_dongle with a incoming context like this:\r\n\r\n```\r\n[mobile-incoming]\r\nexten => _X.,1,Verbose(call from umts)\r\nexten => _X.,n,Playback(demo-echotest)\r\nexten => _X.,n,Echo\r\nexten => _X.,n,Playback(demo-echodone)\r\nexten => _X.,n,Hangup\r\n\r\nexten => _+X.,1,Verbose(call from umts)\r\nexten => _+X.,n,Playback(demo-echotest)\r\nexten => _+X.,n,Echo\r\nexten => _+X.,n,Playback(demo-echodone)\r\nexten => _+X.,n,Hangup\r\n```\r\n\r\nYou need some more packages for playback of the standard asterisk gsm audio files.\r\n\r\nAnd call the dongle. Expected would be to hear the welcome message and then enter the echotest. The welcome message is not understandable on the mobile device also the echotest.\r\n\r\nMy first guess was, that there is a endianess problem (UMTS-stick has little endian, host has big endian) but a quick look into the sourcecode revealed that endianess switching is done on the outgoing channel. The audio sounds like that there is a endianess mixup because calling with a SIP device into the GSM network generates very loud noise on the GSM side while sending silence.\r\n\r\nThe defect looks like\r\n\r\n​https://code.google.com/p/asterisk-chan-dongle/issues/detail?id=112\r\n\r\nbut it's not a codec conversion issue; even sending SLIN-codec from an other box via IAX2 didn't resolve the issue.\r\n\r\nAny idea?\r\n\r\nLeif"},"comment":{"url":"https://api.github.com/repos/openwrt/telephony/issues/comments/77683644","html_url":"https://github.com/openwrt/telephony/issues/7#issuecomment-77683644","issue_url":"https://api.github.com/repos/openwrt/telephony/issues/7","id":77683644,"user":{"login":"andr2k","id":922393,"avatar_url":"https://avatars.githubusercontent.com/u/922393?v=3","gravatar_id":"","url":"https://api.github.com/users/andr2k","html_url":"https://github.com/andr2k","followers_url":"https://api.github.com/users/andr2k/followers","following_url":"https://api.github.com/users/andr2k/following{/other_user}","gists_url":"https://api.github.com/users/andr2k/gists{/gist_id}","starred_url":"https://api.github.com/users/andr2k/starred{/owner}{/repo}","subscriptions_url":"https://api.github.com/users/andr2k/subscriptions","organizations_url":"https://api.github.com/users/andr2k/orgs","repos_url":"https://api.github.com/users/andr2k/repos","events_url":"https://api.github.com/users/andr2k/events{/privacy}","received_events_url":"https://api.github.com/users/andr2k/received_events","type":"User","site_admin":false},"created_at":"2015-03-07T10:42:46Z","updated_at":"2015-03-07T10:42:46Z","body":"Installed fresh version of OpenWRT, latest versions of packages - the issue\nis still here. Tried all  timing modules with the same result:\n- when any of the modules is used, I hear noise instead voice in [network\n-> dongle] direction\n- when they are not used, I hear the voice, but its quality is really\n\n[dongle -> network] voice quality is perfect in both cases.\n\nThe following timing modules were tried: pthread, dahdi, timerfd.\n\nProbably, it is possible to solve the issue without using the modules. Jiri\nSlachta, can you please explain how exactly you played with codecs?\n\nOpenWrt*CLI> core show channeltype Dongle\n-- Info about channel driver: Dongle --\n  Device State: yes\n    Indication: yes\n     Transfer : no\n  Capabilities: (slin)\n   Digit Begin: yes\n     Digit End: yes\n    Send HTML : no\n Image Support: no\n  Text Support: no\nOpenWrt*CLI> core show codecs\nDisclaimer: this command is for informational purposes only.\nIt does not indicate anything about your configuration.\n      ID  TYPE     NAME DESCRIPTION\n-----------------------------------------------------------------------------------\n  100001 audio     g723 (G.723.1)\n  100002 audio      gsm (GSM)\n  100003 audio     ulaw (G.711 u-law)\n  100004 audio     alaw (G.711 A-law)\n  100011 audio     g726 (G.726 RFC3551)\n  100006 audio    adpcm (ADPCM)\n  100019 audio     slin (16 bit Signed Linear PCM)\n  100007 audio    lpc10 (LPC10)\n  100008 audio     g729 (G.729A)\n  100009 audio    speex (SpeeX)\n  100016 audio  speex16 (SpeeX 16khz)\n  100010 audio     ilbc (iLBC)\n  100005 audio g726aal2 (G.726 AAL2)\n  100012 audio     g722 (G722)\n  100021 audio   slin16 (16 bit Signed Linear PCM (16kHz))\n  300001 image     jpeg (JPEG image)\n  300002 image      png (PNG image)\n  200001 video     h261 (H.261 Video)\n  200002 video     h263 (H.263 Video)\n  200003 video    h263p (H.263+ Video)\n  200004 video     h264 (H.264 Video)\n  200005 video    mpeg4 (MPEG4 Video)\n  400001  text      red (T.140 Realtime Text with redundancy)\n  400002  text     t140 (Passthrough T.140 Realtime Text)\n  100013 audio   siren7 (ITU G.722.1 (Siren7, licensed from Polycom))\n  100014 audio  siren14 (ITU G.722.1 Annex C, (Siren14, licensed from\nPolycom))\n  100017 audio  testlaw (G.711 test-law)\n  100015 audio     g719 (ITU G.719)\n  100028 audio  speex32 (SpeeX 32khz)\n  100020 audio   slin12 (16 bit Signed Linear PCM (12kHz))\n  100022 audio   slin24 (16 bit Signed Linear PCM (24kHz))\n  100023 audio   slin32 (16 bit Signed Linear PCM (32kHz))\n  100024 audio   slin44 (16 bit Signed Linear PCM (44kHz))\n  100025 audio   slin48 (16 bit Signed Linear PCM (48kHz))\n  100026 audio   slin96 (16 bit Signed Linear PCM (96kHz))\n  100027 audio  slin192 (16 bit Signed Linear PCM (192kHz))\n\n2015-03-02 16:57 GMT+02:00 iamnague <notifications@github.com>:\n\n> Just an update to tell you that I was not using the correct DATA port\n> (/dev/ttyUSB0 instead of /dev/ttyUSB2) with the dongle. With the correct\n> DATA port, the one way audio issue is not present anymore (with SIP).\n>\n> --\n> Reply to this email directly or view it on GitHub\n> <https://github.com/openwrt/telephony/issues/7#issuecomment-76725689>.\n>\n\n\n\n-- \nAndriy Kravchenko\n\nPhone: +380 96 607 18 75\nE-mail: a@livebb.ru\n"}}