Hi everyone, I am running asterisk on red hat linux 9 box. The sound card is Intel 82801db AC' 97 audio and the module is i810_audio. It runs well with other applications like xmms and the standard tests deliver a sound . I have also tried to record voice and that works well too. 1-)Now when i run asterisk and i dial out an extension to play any sound there is none. The same thing happens when i use sjphone to connect directly to fwd...no ringing sound and no audio ...i could understand that may be a nat issue as greg & steven told me but asterisk on this box and sjphone on this box and no dialing sound ..i have done an iptables -x -f anyhow. I have tried both oss and alsa in modules.conf but nothing worked. I went through the archives and other resources but could get no help. the modules loaded by asterisk are : ######### cdr_pgsql.so PostgreSQL CDR Backend 0 cdr_odbc.so ODBC CDR Backend 0 cdr_csv.so Comma Separated Values CDR Backend 0 format_jpeg.so JPEG (Joint Picture Experts Group) Image 0 format_h263.so Raw h263 data 0 format_pcm_alaw.so Raw aLaw 8khz PCM Audio support 0 format_g729.so Raw G729 data 0 format_pcm.so Raw uLaw 8khz Audio support (PCM) 0 format_vox.so Dialogic VOX (ADPCM) File Format 0 format_wav_gsm.so Microsoft WAV format (Proprietary GSM) 0 format_wav.so Microsoft WAV format (8000hz Signed Line 0 format_gsm.so Raw GSM data 0 codec_a_mu.so A-law and Mulaw direct Coder/Decoder 0 codec_alaw.so A-law Coder/Decoder 0 codec_ulaw.so Mu-law Coder/Decoder 0 codec_adpcm.so Adaptive Differential PCM Coder/Decoder 0 codec_lpc10.so LPC10 2.4kbps (signed linear) Voice Code 0 codec_gsm.so GSM/PCM16 (signed linear) Codec Translat 0 codec_ilbc.so iLBC/PCM16 (signed linear) Codec Transla 0 app_zapscan.so Scan Zap channels application 0 app_zapbarge.so Barge in on Zap channel application 0 app_flash.so Flash zap trunk application 0 app_meetme.so Simple MeetMe conference bridge 0 app_zapras.so Zap RAS Application 0 app_random.so Random goto 0 app_setcdruserfield.so CDR user field apps 0 app_read.so Read Variable Application 0 app_cut.so Cuts up variables 0 app_sayunixtime.so Say time 0 app_hasnewvoicemail.so Indicator for whether a voice mailbox ha 0 app_cdr.so Make sure asterisk doesn't save CDR for 0 app_setcidnum.so Set CallerID Number 0 app_transfer.so Transfer 0 app_enumlookup.so ENUM Lookup 0 app_chanisavail.so Check if channel is available 0 app_db.so Database access functions for Asterisk e 0 app_privacy.so Require phone number to be entered, if n 0 app_waitforring.so Waits until first ring after time 0 app_lookupblacklist.so Look up Caller*ID name/number from black 0 app_softhangup.so Hangs up the requested channel 0 app_authenticate.so Authentication Application 0 app_macro.so Extension Macros 0 app_substring.so Save substring digits in a given variabl 0 app_lookupcidname.so Look up CallerID Name from local databas 0 app_setcidname.so Set CallerID Name 0 app_striplsd.so Strip trailing digits 0 app_parkandannounce.so Call Parking and Announce Application 0 app_senddtmf.so Send DTMF digits Application 0 app_queue.so True Call Queueing 0 app_festival.so Simple Festival Interface 0 app_setcallerid.so Set CallerID Application 0 app_datetime.so Date and Time 0 app_zapateller.so Block Telemarketers with Special Informa 0 app_milliwatt.so Digital Milliwatt (mu-law) Test Applicat 0 app_getcpeid.so Get ADSI CPE ID 0 app_adsiprog.so Asterisk ADSI Programming Application 0 app_qcall.so Call from Queue 0 app_agi.so Asterisk Gateway Interface (AGI) 0 app_disa.so DISA (Direct Inward System Access) Appli 0 app_url.so Send URL Applications 0 app_image.so Image Transmission Application 0 app_record.so Trivial Record Application 0 app_echo.so Simple Echo Application 0 app_system.so Generic System() application 0 app_mp3.so Silly MP3 Application 0 app_directory.so Extension Directory 0 app_voicemail.so Comedian Mail (Voicemail System) 0 app_playback.so Trivial Playback Application 0 app_dial.so Dialing Application 0 pbx_spool.so Outgoing Spool Support 1 pbx_wilcalu.so Wil Cal U (Auto Dialer) 0 pbx_config.so Text Extension Configuration 0 chan_zap.so Zapata Telephony w/PRI 0 chan_phone.so Linux Telephony API Support 0 chan_skinny.so Skinny Client Control Protocol (Skinny) 0 chan_local.so Local Proxy Channel 0 chan_iax2.so Inter Asterisk eXchange (Ver 2) 0 chan_mgcp.so Media Gateway Control Protocol (MGCP) 0 chan_agent.so Agent Proxy Channel 0 chan_modem_i4l.so ISDN4Linux Emulated Modem Driver 0 chan_modem_bestdata.so BestData (Conexant V.90 Chipset) VoiceMo 0 chan_sip.so Session Initiation Protocol (SIP) 0 res_monitor.so Call Monitoring Resource 1 res_indications.so Indications Configuration 0 res_crypto.so Cryptographic Digital Signatures 1 res_parking.so Call Parking Resource 1 res_adsi.so ADSI Resource 1 res_musiconhold.so Music On Hold Resource 1 chan_modem_aopen.so A/Open (Rockwell Chipset) ITU-2 VoiceMod 0 chan_modem.so Generic Voice Modem Driver 0 ############################### the lsmod on the box gives @@@@@@@@@@@@@@@@@@@ zaptel 178144 0 i810_audio 27720 1 (autoclean) ac97_codec 13640 0 (autoclean) [i810_audio] soundcore 6404 2 (autoclean) [i810_audio] i830 74336 1 agpgart 47776 12 (autoclean) parport_pc 19076 1 (autoclean) lp 8996 0 (autoclean) parport 37056 1 (autoclean) [parport_pc lp] autofs 13268 0 (autoclean) (unused) fealnx 13648 1 mii 3976 0 [fealnx] ipt_REJECT 3928 0 (autoclean) iptable_filter 2412 0 (autoclean) ip_tables 15096 2 [ipt_REJECT iptable_filter] sg 36524 0 (autoclean) sr_mod 18136 0 (autoclean) microcode 4668 0 (autoclean) ide-scsi 12208 0 scsi_mod 107160 3 [sg sr_mod ide-scsi] ide-cd 35708 0 cdrom 33728 0 [sr_mod ide-cd] keybdev 2944 0 (unused) mousedev 5492 1 hid 22148 0 (unused) input 5856 0 [keybdev mousedev hid] usb-uhci 26348 0 (unused) ehci-hcd 19976 0 (unused) usbcore 78784 1 [hid usb-uhci ehci-hcd] ext3 70784 2 jbd 51892 2 [ext3] @@@@@@@@@@@@@@@@@@@@@@@@@@ 2-) Imagine i have a static ip of 10.12.x.x and my gateway is 10.12.x.y I have the following settings for sjphone : host 10.12.x.x port 5060 domain 10.12.x.x registrar 10.12.x.x Now whenever i start sjphone and then asterisk it says service unavailable( i did initialize it with the appropriate host secret entries in sip.conf). The same sjphone can dial out to fwd ( of course no sound and with a different setting). Upon sip peers the entry is shown for sjphone . i keep getting a message ################### chan_sip.c:457 __sip_xmit: sip_xmit of 0x80ee78c (len 363) to 192.246.69.223 returned -1: Bad file descriptor Aug 10 18:56:00 WARNING[1116941120]: chan_sip.c:497 retrans_pkt: Maximum retries exceeded on call 6b8b4567327b23c6643c986966334873@10.12.0.5 for seqno 123 (Request) ####################### this message is not sent when i just start asterisk. sometimes i can dial from cli and on other times it says no such command ...... Any help is appreciated..i am in a jungle seemingly and badly need help from yu guys. thanks a tonne niko _________________________________________________________________ Studies, career, romance. Whatever your concerns. http://www.astroyogi.com/newMSN/ We have the answers.
On Sun, 2004-08-08 at 13:29 +0000, niko singh wrote:> Hi everyone, > I am running asterisk on red hat linux 9 box. The sound card is Intel > 82801db AC' 97 audio and the module is i810_audio. It runs well with other > applications like xmms and the standard tests deliver a sound . I have also > tried to record voice and that works well too.Therefore you are running a windows manager, yes.> 1-)Now when i run asterisk and i dial out an extension to play any sound > there is none.On the same machine?> The same thing happens when i use sjphone to connect directly > to fwd...no ringing sound and no audioAsterisk is still running?> ...i could understand that may be a > nat issue as greg & steven told me but asterisk on this box and sjphone on > this box and no dialing sound ..i have done an iptables -x -f anyhow. I have > tried both oss and alsa in modules.conf but nothing worked. I went through > the archives and other resources but could get no help.Mainly because what you are trying is a no no, Asterisk and X certainly with a manager like KDE do not mix well. KISS, use an old low powered PC to provide the front end, I've run 200Mhz PIIs, it will handle iptables and let you experiment with *. Your other machine can then just run X and a window manager and SJPhone, Kphone, Linphone etc. -- Dave Cotton <dcotton@linuxautrement.com>