I am running version 1.4.x. Where do I get PRICAUSE? I tried making a call that was not answered and I did not see any more information. The dumpchan of DADHI/23-1 did not happen as that is in a macro that only gets called for an answered call.
I only see this: Executing [91112223...@empl:8] Dial("SIP/mine-00000521", "Dahdi/G1/1112223333|60|gM(out-dial)") in new stack DEBUG[4907]: dsp.c:1682 ast_dsp_set_busy_pattern: dsp busy pattern set to 0,0 -- Requested transfer capability: 0x00 - SPEECH -- Called G1/1112223333 DEBUG[3188]: chan_dahdi.c:10135 pri_dchannel: Queuing frame from PRI_EVENT_PROCEEDING on channel 0/23 span 1 -- DAHDI/23-1 is proceeding passing it to SIP/mine-00000521 -- DAHDI/23-1 is ringing DEBUG[3188]: chan_dahdi.c:1790 dahdi_enable_ec: Echo cancellation already on -- DAHDI/23-1 answered SIP/mine-00000521 -- Executing [...@macro-out-dial:1] DumpChan("DAHDI/23-1", "") in new stack Dumping Info For Channel: DAHDI/23-1: ================================================================================ Info: Name= DAHDI/23-1 Type= DAHDI UniqueID= sys.domain.com-1294514614.2630 CallerID= 91112223333 CallerIDName= (N/A) DNIDDigits= (N/A) RDNIS= (N/A) State= Up (6) Rings= 0 NativeFormat= 0x4 (ulaw) WriteFormat= 0x4 (ulaw) ReadFormat= 0x4 (ulaw) 1stFileDescriptor= 35 Framesin= 189 Framesout= 176 TimetoHangup= 0 ElapsedTime= 0h0m4s Context= macro-out-dial Extension= s Priority= 1 CallGroup= PickupGroup= Application= DumpChan Data= (Empty) Blocking_in= (Not Blocking) Variables: MACRO_DEPTH=1 MACRO_PRIORITY=1 MACRO_CONTEXT=from-outside MACRO_EXTEN= DIALEDPEERNUMBER=G1/1112223333 TRANSFERCAPABILITY=SPEECH ================================================================================ DEBUG[4907]: app_macro.c:379 _macro_exec: Executed application: DumpChan DEBUG[4907]: app_dial.c:1927 dial_exec_full: Macro exited with status 0 DEBUG[4907]: chan_dahdi.c:3464 dahdi_setoption: Set option AUDIO MODE, value: ON(1) on DAHDI/23-1 DEBUG[4907]: chan_dahdi.c:3092 dahdi_hangup: Not yet hungup... Calling hangup once with icause, and clearing call DEBUG[4907]: chan_dahdi.c:3460 dahdi_setoption: Set option AUDIO MODE, value: OFF(0) on DAHDI/23-1 -- Hungup 'DAHDI/23-1' == Spawn extension (empl, 91112223333, 8) exited non-zero on 'SIP/mine-00000521' -- Executing [...@empl:1] Verbose("SIP/mine-00000521", "2|Hangup SIP/mine-00000521 with cause 16") in new stack == Hangup SIP/mine-00000521 with cause 16 -- Executing [...@empl:2] DumpChan("SIP/mine-00000521", "") in new stack Dumping Info For Channel: SIP/mine-00000521: ================================================================================ Info: Name= SIP/mine-00000521 Type= SIP UniqueID= sys.domain.com-1294514614.2629 CallerID= 4445556666 CallerIDName= Jim Dickenson DNIDDigits= 91112223333 RDNIS= (N/A) State= Up (6) Rings= 0 NativeFormat= 0x2 (gsm) WriteFormat= 0x2 (gsm) ReadFormat= 0x2 (gsm) 1stFileDescriptor= 65 Framesin= 248 Framesout= 253 TimetoHangup= 0 ElapsedTime= 0h0m0s Context= empl Extension= h Priority= 2 CallGroup= PickupGroup= Application= DumpChan Data= (Empty) Blocking_in= (Not Blocking) Variables: DIALSTATUS=ANSWER DIALEDTIME=5 ANSWEREDTIME=1 RTPAUDIOQOS=ssrc=671389293;themssrc=651772178;lp=0;rxjitter=0.001217;rxcount=248;txjitter=0.000000;txcount=252;rlp=0;rtt=0.000000 BRIDGEPEER=DAHDI/23-1 DIALEDPEERNUMBER=G1/1112223333 DIALEDPEERNAME=DAHDI/23-1 MACRO_DEPTH=0 RCStatus=0 MyChan=SIP sipcallid=0b69233cd5469...@192.168.0.16 SIPUSERAGENT=Grandstream GXP2000 1.2.2.6 SIPDOMAIN=sys.domain.com SIPURI=sip:m...@00.00.000.000:5064 ================================================================================ -- Executing [...@empl:3] ExecIf("SIP/mine-00000521", "0|Set|DB(conf//haveadmin)=no") in new stack -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Jan 7, 2011, at 12:44 PM, C F wrote: > PRICAUSE will give you lots of info on why a call was hungup on. Not > sure if SIP will give you the same. > > On Thu, Jan 6, 2011 at 9:06 AM, Jim Dickenson <dicken...@cfmc.com> wrote: >> Does Asterisk, currently using version 1.4, get any more information about >> the result of an outbound call made over a PRI line compared to a call via a >> SIP trunk? >> >> As an example, in a PRI call there is this message that shows up on the >> console: >> >> [2011-01-05 14:59:02] -- Channel 23 detected a CED tone from the network. >> >> for a call to a fax machine. Does asterisk set anything that a dialplan can >> access that can know the call was to a fax machine? >> >> If a call is placed to a number that is disconnected so a special >> information tone is played can either a PRI call or a SIP call know this >> without analyzing the audio stream? >> >> Are there reasons to prefer the use of PRI over SIP or SIP over PRI? >> >> I would like people's opinions as to if one form is better than the other in >> any meaningful way. >> >> Thanks for you feed-back. >> -- >> Jim Dickenson >> mailto:dicken...@cfmc.com >> >> CfMC >> http://www.cfmc.com/ >> >> >> >> >> -- >> _____________________________________________________________________ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> New to Asterisk? 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