additional info on the system Linux home 2.6.30.3-SLACKWARE #1 Sun Feb 7 09:09:33 MYT 2010 i686 Intel(R) Pentium(R) 4 CPU 2.00GHz GenuineIntel GNU/Linux
Asterisk 1.6.2.5, Copyright (C) 1999 - 2009 Digium, Inc. and others. Created by Mark Spencer <marks...@digium.com> Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'core show license' for details. ========================================================================= == Parsing '/etc/asterisk/asterisk.conf': == Found == Parsing '/etc/asterisk/extconfig.conf': == Found Connected to Asterisk 1.6.2.5 currently running on home (pid = 11838) Verbosity is at least 7 home*CLI> module show like dahdi Module Description Use Count codec_dahdi Generic DAHDI Transcoder Codec Translato 0 app_dahdibarge.so Barge in on DAHDI channel application 0 chan_dahdi.so DAHDI Telephony Driver 0 app_dahdiscan.so Scan DAHDI channels application 0 app_dahdiras.so DAHDI ISDN Remote Access Server 0 res_timing_dahdi.so DAHDI Timing Interface 0 on the other hand, calls made internally are ok. On Thu, Mar 4, 2010 at 2:43 PM, Siti Zalifah Md Yatim <ctzali...@gmail.com> wrote: > Hello, > > I'm facing problem where as whenever there are incoming call from > pstn, there will be no audio coming in. User at the other end also > could not hear my voice. This happens few days back. Im using asterisk > 1.6.1.2 with dahdi tool 2.2.0. > > I thought it was time to upgrade, so upgraded to dahdi 2.2.1 and > asterisk 1.6.2.5. However, it does not help at all. > > My current config as follows :- > > X100P clone card > > /etc/dahdi/system.conf > # Span 1: WCFXO/0 "Generic Clone Board 1" (MASTER) > fxsks=1 > echocanceller=mg2,1 > > > /etc/asterisk/dahdi-channels.conf > ; Span 1: WCFXO/0 "Generic Clone Board 1" (MASTER) > ;;; line="1 WCFXO/0/0 FXSKS (SWEC: MG2)" > signalling=fxs_ls > callerid=asreceived > group=0 > context=from-pstn > channel => 1 > callerid= > group= > context=default > > > /etc/asterisk/chan_dahdi.conf > > [trunkgroups] > > > > > [channels] > language = my > ; > usecallerid = yes > callwaiting = yes > usecallingpres = yes > callwaitingcallerid = yes > threewaycalling = yes > transfer = yes > canpark = yes > cancallforward = yes > callreturn = yes > mailbox = 5000 > echocancel = yes > echocancelwhenbridged = yes > rxgain = 2.0 > txgain = 3.0 > group = 1 > callgroup = 1 > pickupgroup = 1 > faxdetect = both > signalling = fxs_ls > callerid = asreceived > group = 0 > channel = 1 > callerid = > group = > context = default > #include "dahdi-channels.conf" > > > my call plan will execute voicemail when there;s incoming call from > pstn. result as shwon here > > > -- Executing [...@from-pstn:1] Set("DAHDI/1-1", "CallTime=20100304 > 13:45:30") in new stack > -- Executing [...@from-pstn:2] Set("DAHDI/1-1", "CallerIDString="" > <01935xxxxx>") in new stack > -- Executing [...@from-pstn:3] System("DAHDI/1-1", "/bin/echo "20100304 > 13:45:30 01935xxxxx [] - to pstn" >> /var/log/asterisk/call_log") in > new stack > -- Executing [...@from-pstn:4] Answer("DAHDI/1-1", "") in new stack > -- Executing [...@from-pstn:5] VoiceMail("DAHDI/1-1", "5000,u") in new stack > -- Stopped music on hold on DAHDI/1-1 > -- Playing 'vm-theperson.gsm' (language 'my') > -- Playing 'digits/5.gsm' (language 'my') > -- Playing 'digits/0.gsm' (language 'my') > -- Playing 'digits/0.gsm' (language 'my') > -- Playing 'digits/0.gsm' (language 'my') > -- Playing 'vm-isunavail.gsm' (language 'my') > -- Playing 'vm-intro.gsm' (language 'my') > -- Playing 'beep.gsm' (language 'my') > -- Recording the message > -- x=0, open writing: > /var/spool/asterisk/voicemail/default/5000/tmp/ksOvKw format: wav, > 0x91bfb68 > -- Recording automatically stopped after a silence of 10 seconds > -- Playing 'auth-thankyou.gsm' (language 'my') > -- Executing [...@from-pstn:8] Hangup("DAHDI/1-1", "") in new stack > > how ever, > starting from line 5 onwards, theres no audio at all. > > anybody can help ? > > thank you. > -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users