[Asterisk-Users] No Sound and Jungle:
niko singh
sharp_in1008 at hotmail.com
Sun Aug 8 06:29:10 MST 2004
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 at 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
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