[RESOLVED] soound problem after config TDM04B


#1

I have a sound problem with TDM04B .after i install the card to PCI and install a@h(v.2.3) as a server , i configure a trunk to use post card and i can call any no i want but the problem is no sound streaming when the other side answer the call. what can i do? please give my any advice to try.


#2

first off, read the sticky at the top of the board. we need configs !! and details. what does “no sound” mean ? nothing at all ?

is it only TDM-routed calls that have no sound ? is the same on all 4 modules ? where are you ? have you loaded the card driver for your locale ? have you adjusted the gain settings ? run fxotune ? monitored the channels while a call is in place ?


#3

first of all thanks very much for ur reply and sorry for no details
this is what Asterisk says when i try to us the trunk:
– Executing Macro(“SIP/15-be6f”, “dialout-trunk|2|0912831331|”) in new stack
– Executing GotoIf(“SIP/15-be6f”, “1?3:2)”) in new stack
– Goto (macro-dialout-trunk,s,3)
– Executing Macro(“SIP/15-be6f”, “user-callerid”) in new stack
– Executing DBget(“SIP/15-be6f”, “AMPUSER=DEVICE/15/user”) in new stack
– DBget: varname=AMPUSER, family=DEVICE, key=15/user
– DBget: set variable AMPUSER to 15
– Executing DBget(“SIP/15-be6f”, “AMPUSERCIDNAME=AMPUSER/15/cidname”) in new stack
– DBget: varname=AMPUSERCIDNAME, family=AMPUSER, key=15/cidname
– DBget: set variable AMPUSERCIDNAME to mohammed
– Executing GotoIf(“SIP/15-be6f”, “0?5”) in new stack
– Executing SetCallerID(“SIP/15-be6f”, ““mohammed” <15>”) in new stack
– Executing NoOp(“SIP/15-be6f”, “Using CallerID “mohammed” <15>”) in new stack
– Executing Macro(“SIP/15-be6f”, “record-enable|15|OUT”) in new stack
– Executing GotoIf(“SIP/15-be6f”, “0 > 0?2:4”) in new stack
– Goto (macro-record-enable,s,4)
– Executing AGI(“SIP/15-be6f”, “recordingcheck|20060206-192205|1139242925.46”) in new stack
– Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
recordingcheck|20060206-192205|1139242925.46: Outbound recording not enabled
– AGI Script recordingcheck completed, returning 0
– Executing NoOp(“SIP/15-be6f”, “No recording needed”) in new stack
– Executing Macro(“SIP/15-be6f”, “outbound-callerid|2”) in new stack
– Executing DBget(“SIP/15-be6f”, “USEROUTCID=AMPUSER/15/outboundcid”) in new stack
– DBget: varname=USEROUTCID, family=AMPUSER, key=15/outboundcid
– DBget: set variable USEROUTCID to
– Executing GotoIf(“SIP/15-be6f”, “1?4”) in new stack
– Goto (macro-outbound-callerid,s,4)
– Executing GotoIf(“SIP/15-be6f”, “1?6”) in new stack
– Goto (macro-outbound-callerid,s,6)
– Executing NoOp(“SIP/15-be6f”, “CallerID set to “mohammed” <15>”) in new stack
– Executing SetGroup(“SIP/15-be6f”, “OUT_2”) in new stack
– Executing CheckGroup(“SIP/15-be6f”, “1”) in new stack
– Executing SetVar(“SIP/15-be6f”, “DIAL_NUMBER=0912831331”) in new stack
– Executing SetVar(“SIP/15-be6f”, “DIAL_TRUNK=2”) in new stack
– Executing AGI(“SIP/15-be6f”, “fixlocalprefix”) in new stack
– Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix
– AGI Script fixlocalprefix completed, returning 0
– Executing SetVar(“SIP/15-be6f”, “OUTNUM=0912831331”) in new stack
– Executing Cut(“SIP/15-be6f”, “custom=OUT_2|:|1”) in new stack
– Executing GotoIf(“SIP/15-be6f”, “0?16”) in new stack
– Executing Dial(“SIP/15-be6f”, “ZAP/1/0912831331”) in new stack
– Called 1/0912831331
– Zap/1-1 answered SIP/15-be6f
– Hungup ‘Zap/1-1’
== Spawn extension (macro-dialout-trunk, s, 14) exited non-zero on ‘SIP/15-be6f’ in macro ‘dialout-trunk’
== Spawn extension (from-internal, 1240912831331, 1) exited non-zero on ‘SIP/15-be6f’
– Executing Macro(“SIP/15-be6f”, “hangupcall”) in new stack
– Executing ResetCDR(“SIP/15-be6f”, “w”) in new stack
– Executing NoCDR(“SIP/15-be6f”, “”) in new stack
– Executing Wait(“SIP/15-be6f”, “5”) in new stack
== Spawn extension (macro-hangupcall, s, 3) exited non-zero on ‘SIP/15-be6f’ in macro ‘hangupcall’
== Spawn extension (from-internal, h, 1) exited non-zero on ‘SIP/15-be6f’

no sound at all in the 2 sides and about the config files i dont modify any of them just i install a@h and configure a trunk to TDM card(i have the same response in 4 modules)

the other routes is work propply and no problem at all
i try to modify the volume after ur reply (i made rxgain=2.0 & txgain=2.0) but nothing new happaned but i dont know how to(runfxotune) or (monitored the channels while a call is in place)
Iam in Ithupia in Africa

thanks again


#4

I had the same problem, I updated to a newer version and that fixed it.


#5

thank u very very much shownpro the problem solved by ur advice I get the v2.5 and reinstall the server and every thing is OK now
Thanks again