[asterisk-users] Benefit of PRI vs SIP trunk calls
Jim Dickenson
dickenson at cfmc.com
Sat Jan 8 19:39:21 CST 2011
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 [91112223333 at 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 [s at 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 [h at 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 [h at 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=0b69233cd5469e06 at 192.168.0.16
SIPUSERAGENT=Grandstream GXP2000 1.2.2.6
SIPDOMAIN=sys.domain.com
SIPURI=sip:mine at 00.00.000.000:5064
================================================================================
-- Executing [h at empl:3] ExecIf("SIP/mine-00000521", "0|Set|DB(conf//haveadmin)=no") in new stack
--
Jim Dickenson
mailto:dickenson at 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 <dickenson at 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:dickenson at cfmc.com
>>
>> CfMC
>> http://www.cfmc.com/
>>
>>
>>
>>
>> --
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>
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