[asterisk-users] HANGUPCAUSE() not working in PJSIP for failed calls
Hi,
This is using Asterisk certified/13.21-cert2, FWIW.
I have a hangup handler on an outgoing SIP channel that grabs the SIP status
like this:
NoOp(keys=${HANGUPCAUSE_KEYS()} sipmsg=${HANGUPCAUSE(${CHANNEL},tech)})
This works fine if the call connects to the other end but the caller for
example hangs up while it's still ringing:
NoOp("PJSIP/custom-00ab", "keys=PJSIP/custom-00ab sipmsg=SIP 180
Ringing") in new stack
It also works fine if the call was answered:
NoOp("PJSIP/custom-00ad",
"keys=PJSIP/custom-00ad,PJSIP/squiresvi-00ac sipmsg=SIP 200 OK") in new
stack
But if the remote end returns an error status (eg 404) then I get nothing:
func_hangupcause.c:140 hangupcause_read: Unable to find information for
channel PJSIP/custom-00af
-- Executing [sipdirect2@hangup_handlers_egress:6]
NoOp("PJSIP/custom-00af", "keys= sipmsg=") in new stack
Any idea whether this is possible? It works fine in our old Asterisk 11
systems, but on Asterisk 13 with PJSIP I'm getting nowhere with it.
Cheers,
Kingsley.
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[asterisk-users] Hangupcause on DAHDI 2.4.9-svn-r9328 channels - Asterisk 1.4.36
Hello, I face a problem on some dahdi incoming calls. Hardware is Xorcom with Elastix 1.6.2.27/Asterisk 1.4.36/DAHDI 2.4.9-svn-r9328 inside. Setup is 3 incoming BRI (euroisdn), ringing phones are 3xSNOM320, 4xSNOM300 and 4xFXS phones. On this calls, phones are ringing and when picked up, nobody on the other end. Other phones are still ringing and same behavior when trying to pickup. No need to say that *8 doesn't work better. I checked one of those call and found that the callee hanged up before someone picked up the call. What I have in logs (debug) logger.c: -- Channel 0/1, span 3 got hangup, cause 111 rtp.c: Channel 'DAHDI/7-1' has no RTP, not doing anything further app_dial.c: Exiting with DIALSTATUS=CANCEL further logger.c: > Protocol Discriminator: Q.931 (8) len=8 logger.c: > TEI=82 Call Ref: len= 1 (reference 1/0x1) (Sent to originator) logger.c: > Message Type: RELEASE COMPLETE (90) logger.c: > [08 02 81 d1] logger.c: > Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0 Location: Private network serving the local user (1) logger.c: > Ext: 1 Cause: Invalid call reference value (81), class = Invalid message (e.g. parameter out of range) (5) ] Questions are: why are phones continuing to ring after the call had been canceled? What are hangup cause 111 and Invalid message/parameter out of range. Thanks for any hint -- Daniel -- _ -- 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
Re: [asterisk-users] ${HANGUPCAUSE} in CDR
On 8 Feb 2011, at 13:30, Shariq Khan wrote:
> Can i add ${HANGUPCAUSE} in CDR after the Dial command using h extension? I
> want to add the Hangup reason of call in userfield of CDR.
http://www.google.com/search?q=asterisk+hangupcause+cdr
Top result... Should do it
Steve
Steve Howes
SMTP to Google proxy Inc
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Re: [asterisk-users] ${HANGUPCAUSE} in CDR
${HANGUPCAUSE} value is available on h extension.
-Original Message-
From: "Shariq Khan"
Sent: Tuesday, February 8, 2011 8:30am
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Subject: [asterisk-users] ${HANGUPCAUSE} in CDR
Hello Gurus,
Can i add ${HANGUPCAUSE} in CDR after the Dial command using h extension? I
want to add the Hangup reason of call in userfield of CDR.
Regards,
Shariq Khan
0333-3501125
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[asterisk-users] ${HANGUPCAUSE} in CDR
Hello Gurus,
Can i add ${HANGUPCAUSE} in CDR after the Dial command using h extension? I
want to add the Hangup reason of call in userfield of CDR.
Regards,
Shariq Khan
0333-3501125
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Re: [asterisk-users] ${HANGUPCAUSE} is always 0 in the h extension
That is normal.
In any case, if the one of call party is hangup, then you take the
hangupcase 16 - normal call clearing.
But if you want to see who is was terminatet the call, you must look in
h in boot leg of the call, incoming and outgoing. and so you can see who
is hangup the call.
I think you can understand what I have write. In this case I can see who
has hang up call, and the billsec for both side.
This is with FailOver call.
Look my dial plan in ael, I use this with this call out files:
==
Channel: local/calltocusto...@c2c_c2customerviavendors/n
Callerid: 18185402021
WaitTime: 45
MaxRetries: 0
RetryTime: 10
Account: AGENT1
Context: c2c_C2AgentsVIAVendors
Extension: 18185402021
Priority: 1
Setvar: CUSTOMERPEER="ToVendor3-ToVendor4"
Setvar: AGENTPEER="ToVendor1-ToVendor2"
Setvar: CUSTOMERPHONE=37410219709
Setvar: AGENTPHONE=18185402021
Setvar: AGENTTECH=SIP
Setvar: CUSTOMERTECH=SIP
Archive: yes
Dialplan in AEL:
===
context c2c_C2CustomerVIASIP {
_CallToCustomer => {
Dial(SIP/${custom...@${exten});
Hangup;
};
};
context c2c_C2CustomerVIAVendors {
_CallToCustomer => {
CALLERID(all)=${AGENTPHONE}; Noop(${CALLERID(number)});
&FailOverDial(${CUSTOMERTECH},${CUSTOMERPEER},${CUSTOMERPHONE},${CONTEXT});
Noop(CustomerVIAVerdors == return from macro);
Hangup;
};
h => { Noop(Hangup in c2c_C2CustomerVIAVendors );
Noop(: Duration of the call once it was answered.${CDR(billsec)});
};
};
context c2c_C2AgentsVIASIP {
_CallToCustomer => {
Dial(SIP/${age...@${exten});
Hangup;
};
};
context c2c_C2AgentsVIAVendors {
_CallToCustomer => {
CALLERID(all)=${CUSTOMERPHONE}; Noop(${CALLERID(number)});
&FailOverDial(${AGENTTECH},${AGENTPEER},${AGENTPHONE},${CONTEXT});
Noop(AgentsVIAVendors === return from macro);
Hangup;
};
_X. => {
CALLERID(all)=${CUSTOMERPHONE}; Noop(${CALLERID(number)});
&FailOverDial(${AGENTTECH},${AGENTPEER},${AGENTPHONE},${CONTEXT});
Noop(AgentsVIAVendors === return from macro);
Hangup;
};
h => { Noop(Hangup in c2c_AgentsVIAVendors );
Noop(: Duration of the call once it was answered.${CDR(billsec)});
Hangup;
};
};
macro FailOverDial (tech,peers,phone,cont) {
i=1; focount=0;
Set(j=${CUT(peers,,${i})});
leng=${LEN(${j})};
noop(Tech:${tech}); noop(Peers:${peers}); noop(Phone:${phone});
noop(J:${j});
while ( ${leng} > 0 ) {
Noop(${tech}/${pho...@${j});
Dial(${tech}/${pho...@${j});
Noop(Dial status:${DIALSTATUS});
Noop(End dial command);
switch(${DIALSTATUS}) {
case BUSY:
Noop(Busy);
return;
case CHANUNAVAIL:
Noop(Channel unavailble);
i=${i}+1;
Set(j=${CUT(peers,,${i})});
leng=${LEN(${j})};
noop(J:${j});
break;
case NOANSWER:
Noop(No answer);
return;
case CANCEL:
Noop(Cancel);
return;
case CONGESTION:
Noop(Congestion);
i=${i}+1;
Set(j=${CUT(peers,,${i})});
leng=${LEN(${j})};
break;
case ANSWER:
Noop( Answer);
return;
default:
Noop(Defaul);
return;
};
};
catch h {
Noop(Hagup in macro);
Noop(: Duration of the call once it was answered.${CDR(billsec)});
Noop(Hangup for context:${cont});
&ShowCDRDetails();
Hangup;
return;
};
return;
};
macro ShowCDRDetails () {
Noop(Hangup in macro);
Noop(: The channel's account code:${CDR(accountcode)});
Noop(: DOCUMENTATION, BILL, IGNORE etc:${CDR(amaflags)});
Noop(: Time the call was answered:${CDR(answer)});
Noop(: Duration of the call once it was answered.${CDR(billsec)});
Noop(: Channel name:${CDR(channel)});
Noop(: Caller ID:${CDR(clid)});
Noop(: Destination context:${CDR(dcontext)});
Noop(: ANSWERED, NO ANSWER, BUSY:${CDR(disposition)});
Noop(: Destination:${CDR(dst)});
Noop(: Destination channel:${CDR(dstchannel)});
Noop(: Duration of the call:${CDR(duration)});
Noop(: Time the call ended:${CDR(end)});
Noop(: Last app executed:${CDR(lastapp)});
Noop(: Last app's arguments:${CDR(lastdata)});
Noop(: Source:${CDR(src)});
Noop(: Time the call started:${CDR(start)});
Noop(: The channel's unique id:${CDR(uniqueid)});
return;
};
Zhang Shukun wrote:
> this is dialplan:
>
Re: [asterisk-users] ${HANGUPCAUSE} is always 0 in the h extension
this is dialplan:
exten => 123,1,Dial(SIP/1000,10,L(1))
exten => 123,2,NoOp(HANGUPCAUSE is ${HANGUPCAUSE})
this is the log which hangup by caller:
== Using SIP RTP CoS mark 5
-- Executing [...@95040:1] Dial("SIP/1001-0031",
"SIP/1000,10,L(1)") in new stack
-- Setting call duration limit to 10.000 seconds.
== Using SIP RTP CoS mark 5
-- Called 1000
-- SIP/1000-0032 is ringing
-- SIP/1000-0032 answered SIP/1001-0031
-- Executing [...@95040:1] Playback("SIP/1001-0031",
"vm-goodbye") in new stack
[May 11 17:23:16] WARNING[4258]: file.c:750 ast_readaudio_callback:
Failed to write frame
-- Playing 'vm-goodbye.gsm' (language 'en')
[May 11 17:23:16] WARNING[4258]: app_playback.c:471 playback_exec:
ast_streamfile failed on SIP/1001-0031 for vm-goodbye
-- Executing [...@95040:2] NoOp("SIP/1001-0031", "HANGUPCAUSE is
16") in new stack
== Spawn extension (95040, 123, 1) exited non-zero on 'SIP/1001-0031'
this is the log which hangup by callee:
== Using SIP RTP CoS mark 5
-- Executing [...@95040:1] Dial("SIP/1001-0033",
"SIP/1000,10,L(1)") in new stack
-- Setting call duration limit to 10.000 seconds.
== Using SIP RTP CoS mark 5
-- Called 1000
-- SIP/1000-0034 is ringing
-- SIP/1000-0034 answered SIP/1001-0033
-- Executing [...@95040:1] Playback("SIP/1001-0033",
"vm-goodbye") in new stack
-- Playing 'vm-goodbye.gsm' (language 'en')
-- Executing [...@95040:2] NoOp("SIP/1001-0033", "HANGUPCAUSE is
16") in new stack
== Spawn extension (95040, 123, 1) exited non-zero on 'SIP/1001-0033'
2010/5/11 Vardan :
> Can you show your dialplan part for that call and log also please
>
> Thanks
>
> Zhang Shukun wrote:
>> thank you for reply.
>>
>> but hangupcause cant different whether caller hangup or callee hangup?
>>
>> above two situation both return 16.
>>
>> 2010/5/11 Vardan:
>>> Asterisk variable hangupcause
>>> Page Contents
>>>
>>> * Asterisk variable Hangupcause
>>> o Recommended SIP<-> ISDN Cause codes (from RFC3398):
>>> o PRI Hangup Codes
>>> o Version notes
>>> o Tip
>>> o Examples
>>> + Example 1
>>> + Example 2
>>> + Example 3: Macro for handling hangupcause
>>> + Example 4: Set the hangup cause text to a variable
>>> o See also
>>>
>>>
>>> Asterisk variable Hangupcause
>>> Hangupcause is the latest PRI hangup return code on a zap channel
>>> connected to a PRI interface. Note that this also works on SIP channels,
>>> maybe other channels as well.
>>> Tip: The packet isdnutils contains a utility called isdncause that
>>> provides a textual explanation of the error number that you feed it with
>>> (watch the entry format).
>>>
>>> Previous to CVS 2004-08-12:
>>>
>>> From causes.h:
>>> #define AST_CAUSE_NOTDEFINED 0
>>> #define AST_CAUSE_NORMAL 1
>>> #define AST_CAUSE_BUSY 2
>>> #define AST_CAUSE_FAILURE 3
>>> #define AST_CAUSE_CONGESTION 4
>>> #define AST_CAUSE_UNALLOCATED 5
>>>
>>>
>>> For CVS head releases after 2004-08-12:
>>>
>>> /* Causes for disconnection (from Q.931) */
>>> #define AST_CAUSE_UNALLOCATED 1
>>> #define AST_CAUSE_NO_ROUTE_TRANSIT_NET 2
>>> #define AST_CAUSE_NO_ROUTE_DESTINATION 3
>>> #define AST_CAUSE_CHANNEL_UNACCEPTABLE 6
>>> #define AST_CAUSE_CALL_AWARDED_DELIVERED 7
>>> #define AST_CAUSE_NORMAL_CLEARING 16
>>> #define AST_CAUSE_USER_BUSY 17
>>> #define AST_CAUSE_NO_USER_RESPONSE 18
>>> #define AST_CAUSE_NO_ANSWER 19
>>> #define AST_CAUSE_CALL_REJECTED 21
>>> #define AST_CAUSE_NUMBER_CHANGED 22
>>> #define AST_CAUSE_DESTINATION_OUT_OF_ORDER 27
>>> #define AST_CAUSE_INVALID_NUMBER_FORMAT 28
>>> #define AST_CAUSE_FACILITY_REJECTED 29
>>> #define AST_CAUSE_RESPONSE_TO_STATUS_ENQUIRY 30
>>> #define AST_CAUSE_NORMAL_UNSPECIFIED 31
>>> #define AST_CAUSE_NORMAL_CIRCUIT_CONGESTION 34
>>> #define AST_CAUSE_NETWORK_OUT_OF_ORDER 38
>>> #define AST_CAUSE_NORMAL_TEMPORARY_FAILURE 41
>>> #define AST_CAUSE_SWITCH_CONGESTION 42
>>> #define AST_CAUSE_ACCESS_INFO_DISCARDED 43
>>> #define AST_CAUSE_REQUESTED_CHAN_UNAVAIL 44
>>> #define AST_CAUSE_PRE_EMPTED 45
>>> #define AST_CAUSE_FACILITY_NOT_SUBSCRIBED 50
>>> #define AST_CAUSE_OUTGOING_CALL_BARRED 52
>>> #define AST_CAUSE_INCOMING_CALL_BARRED 54
>>> #define AST_CAUSE_BEARERCAPABILITY_NOTAUTH 57
>>> #define AST_CAUSE_BEARERCAPABILITY_NOTAVAIL 58
>>> #define AST_CAUSE_BEARERCAPABILITY_NOTIMPL 65
>>> #define AST_CAUSE_CHAN_NOT_IMPLEMENTED 66
>>> #define AST_CAUSE_FACILITY_NOT_IMPLEMENTED 69
>>> #define AST_CAUSE_INVALID_CALL_REFERENCE 81
>>> #define AST_CAUSE_INCOMPATIBLE_DESTINATION 88
>>> #define AST_CAUSE_INVALID_MSG_UNSPECIFIED 95
>>> #define AST_CAUSE_MANDATORY_IE_MISSING 96
>>> #define AST_CAUSE_MESSAGE_TYPE_NONEXIST 97
>>>
Re: [asterisk-users] ${HANGUPCAUSE} is always 0 in the h extension
Can you show your dialplan part for that call and log also please Thanks Zhang Shukun wrote: > thank you for reply. > > but hangupcause cant different whether caller hangup or callee hangup? > > above two situation both return 16. > > 2010/5/11 Vardan: >> Asterisk variable hangupcause >> Page Contents >> >> * Asterisk variable Hangupcause >>o Recommended SIP<-> ISDN Cause codes (from RFC3398): >>o PRI Hangup Codes >>o Version notes >>o Tip >>o Examples >> + Example 1 >> + Example 2 >> + Example 3: Macro for handling hangupcause >> + Example 4: Set the hangup cause text to a variable >>o See also >> >> >> Asterisk variable Hangupcause >> Hangupcause is the latest PRI hangup return code on a zap channel >> connected to a PRI interface. Note that this also works on SIP channels, >> maybe other channels as well. >> Tip: The packet isdnutils contains a utility called isdncause that >> provides a textual explanation of the error number that you feed it with >> (watch the entry format). >> >> Previous to CVS 2004-08-12: >> >> From causes.h: >> #define AST_CAUSE_NOTDEFINED0 >> #define AST_CAUSE_NORMAL1 >> #define AST_CAUSE_BUSY 2 >> #define AST_CAUSE_FAILURE 3 >> #define AST_CAUSE_CONGESTION4 >> #define AST_CAUSE_UNALLOCATED 5 >> >> >> For CVS head releases after 2004-08-12: >> >> /* Causes for disconnection (from Q.931) */ >> #define AST_CAUSE_UNALLOCATED 1 >> #define AST_CAUSE_NO_ROUTE_TRANSIT_NET 2 >> #define AST_CAUSE_NO_ROUTE_DESTINATION 3 >> #define AST_CAUSE_CHANNEL_UNACCEPTABLE 6 >> #define AST_CAUSE_CALL_AWARDED_DELIVERED 7 >> #define AST_CAUSE_NORMAL_CLEARING 16 >> #define AST_CAUSE_USER_BUSY 17 >> #define AST_CAUSE_NO_USER_RESPONSE 18 >> #define AST_CAUSE_NO_ANSWER 19 >> #define AST_CAUSE_CALL_REJECTED 21 >> #define AST_CAUSE_NUMBER_CHANGED 22 >> #define AST_CAUSE_DESTINATION_OUT_OF_ORDER 27 >> #define AST_CAUSE_INVALID_NUMBER_FORMAT 28 >> #define AST_CAUSE_FACILITY_REJECTED 29 >> #define AST_CAUSE_RESPONSE_TO_STATUS_ENQUIRY 30 >> #define AST_CAUSE_NORMAL_UNSPECIFIED 31 >> #define AST_CAUSE_NORMAL_CIRCUIT_CONGESTION 34 >> #define AST_CAUSE_NETWORK_OUT_OF_ORDER 38 >> #define AST_CAUSE_NORMAL_TEMPORARY_FAILURE 41 >> #define AST_CAUSE_SWITCH_CONGESTION 42 >> #define AST_CAUSE_ACCESS_INFO_DISCARDED 43 >> #define AST_CAUSE_REQUESTED_CHAN_UNAVAIL 44 >> #define AST_CAUSE_PRE_EMPTED 45 >> #define AST_CAUSE_FACILITY_NOT_SUBSCRIBED 50 >> #define AST_CAUSE_OUTGOING_CALL_BARRED 52 >> #define AST_CAUSE_INCOMING_CALL_BARRED 54 >> #define AST_CAUSE_BEARERCAPABILITY_NOTAUTH 57 >> #define AST_CAUSE_BEARERCAPABILITY_NOTAVAIL 58 >> #define AST_CAUSE_BEARERCAPABILITY_NOTIMPL 65 >> #define AST_CAUSE_CHAN_NOT_IMPLEMENTED 66 >> #define AST_CAUSE_FACILITY_NOT_IMPLEMENTED 69 >> #define AST_CAUSE_INVALID_CALL_REFERENCE 81 >> #define AST_CAUSE_INCOMPATIBLE_DESTINATION 88 >> #define AST_CAUSE_INVALID_MSG_UNSPECIFIED 95 >> #define AST_CAUSE_MANDATORY_IE_MISSING 96 >> #define AST_CAUSE_MESSAGE_TYPE_NONEXIST 97 >> #define AST_CAUSE_WRONG_MESSAGE 98 >> #define AST_CAUSE_IE_NONEXIST 99 >> #define AST_CAUSE_INVALID_IE_CONTENTS 100 >> #define AST_CAUSE_WRONG_CALL_STATE 101 >> #define AST_CAUSE_RECOVERY_ON_TIMER_EXPIRE 102 >> #define AST_CAUSE_MANDATORY_IE_LENGTH_ERROR 103 >> #define AST_CAUSE_PROTOCOL_ERROR 111 >> #define AST_CAUSE_INTERWORKING 127 >> /* Special Asterisk aliases */ >> #define AST_CAUSE_BUSY AST_CAUSE_USER_BUSY >> #define AST_CAUSE_FAILURE AST_CAUSE_NETWORK_OUT_OF_ORDER >> #define AST_CAUSE_NORMAL AST_CAUSE_NORMAL_CLEARING >> #define AST_CAUSE_NOANSWER AST_CAUSE_NO_ANSWER >> #define AST_CAUSE_CONGESTION AST_CAUSE_NORMAL_CIRCUIT_CONGESTION >> #define AST_CAUSE_NOTDEFINED 0 >> >> >> >> Note: This does not work in 0.7.1 (maybe other versions) See: >> http://bugs.digium.com/bug_view_page.php?bug_id=890 >> >> Recommended SIP<-> ISDN Cause codes (from RFC3398): >> >>ISUP Cause valueSIP response >> >>1 unallocated number 404 Not Found >>2 no route to network 404 Not found >>3 no route to destination 404 Not found >>16 normal call clearing --- (*) >>17 user busy486 Busy here >>18 no user responding 408 Request Timeout >>19 no answer from the user 480 Temporarily unavailable >>20 subscriber absent480 Temporarily unavailable >>21 call rejected403 Forbidden (+) >>22 number changed (w/o diagnostic) 410 Gone >>22 number changed (w/ diagnostic) 301 Moved Permanently >>23 redirection to new desti
Re: [asterisk-users] ${HANGUPCAUSE} is always 0 in the h extension
thank you for reply. but hangupcause cant different whether caller hangup or callee hangup? above two situation both return 16. 2010/5/11 Vardan : > Asterisk variable hangupcause > Page Contents > > * Asterisk variable Hangupcause > o Recommended SIP <-> ISDN Cause codes (from RFC3398): > o PRI Hangup Codes > o Version notes > o Tip > o Examples > + Example 1 > + Example 2 > + Example 3: Macro for handling hangupcause > + Example 4: Set the hangup cause text to a variable > o See also > > > Asterisk variable Hangupcause > Hangupcause is the latest PRI hangup return code on a zap channel > connected to a PRI interface. Note that this also works on SIP channels, > maybe other channels as well. > Tip: The packet isdnutils contains a utility called isdncause that > provides a textual explanation of the error number that you feed it with > (watch the entry format). > > Previous to CVS 2004-08-12: > > From causes.h: > #define AST_CAUSE_NOTDEFINED 0 > #define AST_CAUSE_NORMAL 1 > #define AST_CAUSE_BUSY 2 > #define AST_CAUSE_FAILURE 3 > #define AST_CAUSE_CONGESTION 4 > #define AST_CAUSE_UNALLOCATED 5 > > > For CVS head releases after 2004-08-12: > > /* Causes for disconnection (from Q.931) */ > #define AST_CAUSE_UNALLOCATED 1 > #define AST_CAUSE_NO_ROUTE_TRANSIT_NET 2 > #define AST_CAUSE_NO_ROUTE_DESTINATION 3 > #define AST_CAUSE_CHANNEL_UNACCEPTABLE 6 > #define AST_CAUSE_CALL_AWARDED_DELIVERED 7 > #define AST_CAUSE_NORMAL_CLEARING 16 > #define AST_CAUSE_USER_BUSY 17 > #define AST_CAUSE_NO_USER_RESPONSE 18 > #define AST_CAUSE_NO_ANSWER 19 > #define AST_CAUSE_CALL_REJECTED 21 > #define AST_CAUSE_NUMBER_CHANGED 22 > #define AST_CAUSE_DESTINATION_OUT_OF_ORDER 27 > #define AST_CAUSE_INVALID_NUMBER_FORMAT 28 > #define AST_CAUSE_FACILITY_REJECTED 29 > #define AST_CAUSE_RESPONSE_TO_STATUS_ENQUIRY 30 > #define AST_CAUSE_NORMAL_UNSPECIFIED 31 > #define AST_CAUSE_NORMAL_CIRCUIT_CONGESTION 34 > #define AST_CAUSE_NETWORK_OUT_OF_ORDER 38 > #define AST_CAUSE_NORMAL_TEMPORARY_FAILURE 41 > #define AST_CAUSE_SWITCH_CONGESTION 42 > #define AST_CAUSE_ACCESS_INFO_DISCARDED 43 > #define AST_CAUSE_REQUESTED_CHAN_UNAVAIL 44 > #define AST_CAUSE_PRE_EMPTED 45 > #define AST_CAUSE_FACILITY_NOT_SUBSCRIBED 50 > #define AST_CAUSE_OUTGOING_CALL_BARRED 52 > #define AST_CAUSE_INCOMING_CALL_BARRED 54 > #define AST_CAUSE_BEARERCAPABILITY_NOTAUTH 57 > #define AST_CAUSE_BEARERCAPABILITY_NOTAVAIL 58 > #define AST_CAUSE_BEARERCAPABILITY_NOTIMPL 65 > #define AST_CAUSE_CHAN_NOT_IMPLEMENTED 66 > #define AST_CAUSE_FACILITY_NOT_IMPLEMENTED 69 > #define AST_CAUSE_INVALID_CALL_REFERENCE 81 > #define AST_CAUSE_INCOMPATIBLE_DESTINATION 88 > #define AST_CAUSE_INVALID_MSG_UNSPECIFIED 95 > #define AST_CAUSE_MANDATORY_IE_MISSING 96 > #define AST_CAUSE_MESSAGE_TYPE_NONEXIST 97 > #define AST_CAUSE_WRONG_MESSAGE 98 > #define AST_CAUSE_IE_NONEXIST 99 > #define AST_CAUSE_INVALID_IE_CONTENTS 100 > #define AST_CAUSE_WRONG_CALL_STATE 101 > #define AST_CAUSE_RECOVERY_ON_TIMER_EXPIRE 102 > #define AST_CAUSE_MANDATORY_IE_LENGTH_ERROR 103 > #define AST_CAUSE_PROTOCOL_ERROR 111 > #define AST_CAUSE_INTERWORKING 127 > /* Special Asterisk aliases */ > #define AST_CAUSE_BUSY AST_CAUSE_USER_BUSY > #define AST_CAUSE_FAILURE AST_CAUSE_NETWORK_OUT_OF_ORDER > #define AST_CAUSE_NORMAL AST_CAUSE_NORMAL_CLEARING > #define AST_CAUSE_NOANSWER AST_CAUSE_NO_ANSWER > #define AST_CAUSE_CONGESTION AST_CAUSE_NORMAL_CIRCUIT_CONGESTION > #define AST_CAUSE_NOTDEFINED 0 > > > > Note: This does not work in 0.7.1 (maybe other versions) See: > http://bugs.digium.com/bug_view_page.php?bug_id=890 > > Recommended SIP <-> ISDN Cause codes (from RFC3398): > > ISUP Cause value SIP response > > 1 unallocated number 404 Not Found > 2 no route to network 404 Not found > 3 no route to destination 404 Not found > 16 normal call clearing --- (*) > 17 user busy 486 Busy here > 18 no user responding 408 Request Timeout > 19 no answer from the user 480 Temporarily unavailable > 20 subscriber absent 480 Temporarily unavailable > 21 call rejected 403 Forbidden (+) > 22 number changed (w/o diagnostic) 410 Gone > 22 number changed (w/ diagnostic) 301 Moved Permanently > 23 redirection to new destination 410 Gone > 26 non-selected user clearing 404 Not Found (=) > 27 destination out of order 502 Bad Gateway > 28 address incomplete 484 Address incomplete > > > Zhang Shukun wrote: >> hi , all >> >> i want to wtite hangupcause to cdr, but
Re: [asterisk-users] ${HANGUPCAUSE} is always 0 in the h extension
Asterisk variable hangupcause
Page Contents
* Asterisk variable Hangupcause
o Recommended SIP <-> ISDN Cause codes (from RFC3398):
o PRI Hangup Codes
o Version notes
o Tip
o Examples
+ Example 1
+ Example 2
+ Example 3: Macro for handling hangupcause
+ Example 4: Set the hangup cause text to a variable
o See also
Asterisk variable Hangupcause
Hangupcause is the latest PRI hangup return code on a zap channel
connected to a PRI interface. Note that this also works on SIP channels,
maybe other channels as well.
Tip: The packet isdnutils contains a utility called isdncause that
provides a textual explanation of the error number that you feed it with
(watch the entry format).
Previous to CVS 2004-08-12:
From causes.h:
#define AST_CAUSE_NOTDEFINED0
#define AST_CAUSE_NORMAL1
#define AST_CAUSE_BUSY 2
#define AST_CAUSE_FAILURE 3
#define AST_CAUSE_CONGESTION4
#define AST_CAUSE_UNALLOCATED 5
For CVS head releases after 2004-08-12:
/* Causes for disconnection (from Q.931) */
#define AST_CAUSE_UNALLOCATED 1
#define AST_CAUSE_NO_ROUTE_TRANSIT_NET 2
#define AST_CAUSE_NO_ROUTE_DESTINATION 3
#define AST_CAUSE_CHANNEL_UNACCEPTABLE 6
#define AST_CAUSE_CALL_AWARDED_DELIVERED 7
#define AST_CAUSE_NORMAL_CLEARING 16
#define AST_CAUSE_USER_BUSY 17
#define AST_CAUSE_NO_USER_RESPONSE 18
#define AST_CAUSE_NO_ANSWER 19
#define AST_CAUSE_CALL_REJECTED 21
#define AST_CAUSE_NUMBER_CHANGED 22
#define AST_CAUSE_DESTINATION_OUT_OF_ORDER 27
#define AST_CAUSE_INVALID_NUMBER_FORMAT 28
#define AST_CAUSE_FACILITY_REJECTED 29
#define AST_CAUSE_RESPONSE_TO_STATUS_ENQUIRY 30
#define AST_CAUSE_NORMAL_UNSPECIFIED 31
#define AST_CAUSE_NORMAL_CIRCUIT_CONGESTION 34
#define AST_CAUSE_NETWORK_OUT_OF_ORDER 38
#define AST_CAUSE_NORMAL_TEMPORARY_FAILURE 41
#define AST_CAUSE_SWITCH_CONGESTION 42
#define AST_CAUSE_ACCESS_INFO_DISCARDED 43
#define AST_CAUSE_REQUESTED_CHAN_UNAVAIL 44
#define AST_CAUSE_PRE_EMPTED 45
#define AST_CAUSE_FACILITY_NOT_SUBSCRIBED 50
#define AST_CAUSE_OUTGOING_CALL_BARRED 52
#define AST_CAUSE_INCOMING_CALL_BARRED 54
#define AST_CAUSE_BEARERCAPABILITY_NOTAUTH 57
#define AST_CAUSE_BEARERCAPABILITY_NOTAVAIL 58
#define AST_CAUSE_BEARERCAPABILITY_NOTIMPL 65
#define AST_CAUSE_CHAN_NOT_IMPLEMENTED 66
#define AST_CAUSE_FACILITY_NOT_IMPLEMENTED 69
#define AST_CAUSE_INVALID_CALL_REFERENCE 81
#define AST_CAUSE_INCOMPATIBLE_DESTINATION 88
#define AST_CAUSE_INVALID_MSG_UNSPECIFIED 95
#define AST_CAUSE_MANDATORY_IE_MISSING 96
#define AST_CAUSE_MESSAGE_TYPE_NONEXIST 97
#define AST_CAUSE_WRONG_MESSAGE 98
#define AST_CAUSE_IE_NONEXIST 99
#define AST_CAUSE_INVALID_IE_CONTENTS 100
#define AST_CAUSE_WRONG_CALL_STATE 101
#define AST_CAUSE_RECOVERY_ON_TIMER_EXPIRE 102
#define AST_CAUSE_MANDATORY_IE_LENGTH_ERROR 103
#define AST_CAUSE_PROTOCOL_ERROR 111
#define AST_CAUSE_INTERWORKING 127
/* Special Asterisk aliases */
#define AST_CAUSE_BUSY AST_CAUSE_USER_BUSY
#define AST_CAUSE_FAILURE AST_CAUSE_NETWORK_OUT_OF_ORDER
#define AST_CAUSE_NORMAL AST_CAUSE_NORMAL_CLEARING
#define AST_CAUSE_NOANSWER AST_CAUSE_NO_ANSWER
#define AST_CAUSE_CONGESTION AST_CAUSE_NORMAL_CIRCUIT_CONGESTION
#define AST_CAUSE_NOTDEFINED 0
Note: This does not work in 0.7.1 (maybe other versions) See:
http://bugs.digium.com/bug_view_page.php?bug_id=890
Recommended SIP <-> ISDN Cause codes (from RFC3398):
ISUP Cause valueSIP response
1 unallocated number 404 Not Found
2 no route to network 404 Not found
3 no route to destination 404 Not found
16 normal call clearing --- (*)
17 user busy486 Busy here
18 no user responding 408 Request Timeout
19 no answer from the user 480 Temporarily unavailable
20 subscriber absent480 Temporarily unavailable
21 call rejected403 Forbidden (+)
22 number changed (w/o diagnostic) 410 Gone
22 number changed (w/ diagnostic) 301 Moved Permanently
23 redirection to new destination 410 Gone
26 non-selected user clearing 404 Not Found (=)
27 destination out of order 502 Bad Gateway
28 address incomplete 484 Address incomplete
Zhang Shukun wrote:
> hi , all
>
> i want to wtite hangupcause to cdr, but both caller hangup and
> callee hangup result in hangupcause code 16.
>
> how would i know whether caller or callee or system error hangup the phone?
>
> please help.
>
> thanks!
>
> 2010/4/22 Alejandro Recarey:
>>> However, as I can see by the verbose command, ${HANGUPCAUSE} is a
Re: [asterisk-users] ${HANGUPCAUSE} is always 0 in the h extension
hi , all
i want to wtite hangupcause to cdr, but both caller hangup and
callee hangup result in hangupcause code 16.
how would i know whether caller or callee or system error hangup the phone?
please help.
thanks!
2010/4/22 Alejandro Recarey :
>> However, as I can see by the verbose command, ${HANGUPCAUSE} is always
>> 0. I thought it was a channel variable that contained the hangupcause?
>
> Just an update, if the call is established, then there is a
> hangupcause received.
>
> The above problem only happens if the caller hangs up before pickup.
>
> This is usualy a cause 16, not 0.
>
> Alex
>
> --
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Thanks for your supporting,
have a nice day.
Sucan
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Re: [asterisk-users] ${HANGUPCAUSE} is always 0 in the h extension
> However, as I can see by the verbose command, ${HANGUPCAUSE} is always
> 0. I thought it was a channel variable that contained the hangupcause?
Just an update, if the call is established, then there is a
hangupcause received.
The above problem only happens if the caller hangs up before pickup.
This is usualy a cause 16, not 0.
Alex
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[asterisk-users] ${HANGUPCAUSE} is always 0 in the h extension
Hi all,
I am using cdr_adaptive_odbc and it works fine. I am trying to save
the q931 hangupcause to a cdr record. My diaplan looks like this.
exten => _X.,1,Dial(${EXTEN})
exten => h,1,Set(CDR(q931)=${HANGUPCAUSE})
exten => h,2,Verbose(${HANGUPCAUSE})
However, as I can see by the verbose command, ${HANGUPCAUSE} is always
0. I thought it was a channel variable that contained the hangupcause?
How can I set this up to correctly save the hangupcause??
Thank you for your help
Regards,
Alex
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[asterisk-users] ${HANGUPCAUSE} is not printed when call ends or is interrupted
Today I get the remark that a call got disconnected after 10 minutes.
This what my VERBOSE-logfile tells me :
[May 18 15:36:30] VERBOSE[3940] logger.c: -- Executing
[00493516...@intern:1] NoOp("SIP/51-b76023b8", "Gesprek naar GSM-nummer
via Telenet") in new stack
[May 18 15:36:30] VERBOSE[3940] logger.c: -- Executing
[00493516...@intern:2] Dial("SIP/51-b76023b8", "DAHDI/g1/0493516426") in
new stack
[May 18 15:36:30] DEBUG[3940] dsp.c: dsp busy pattern set to 0,0
[May 18 15:36:30] DEBUG[3940] chan_dahdi.c: Dialing '0493516426'
[May 18 15:36:30] DEBUG[3940] chan_dahdi.c: Deferring dialing...
[May 18 15:36:30] VERBOSE[3940] logger.c: -- Called g1/0493516426
[May 18 15:36:31] DEBUG[3940] chan_dahdi.c: Sent deferred digit string:
T0493516426w
[May 18 15:36:33] VERBOSE[3940] logger.c: -- DAHDI/1-1 answered
SIP/51-b76023b8
[May 18 15:49:35] VERBOSE[3940] logger.c: -- Hungup 'DAHDI/1-1'
[May 18 15:49:35] VERBOSE[3940] logger.c: == Spawn extension (intern,
00493516426, 2) exited non-zero on 'SIP/51-b76023b8'
Nothing abnormal I think ?!
By the way : is there a way to see which end of the conversation ended
the call ??
If the call is disconnected, I have the following in my dialplan to
debug the cause of an interrupted call :
exten => _00ZXXX,1,NoOp(national conversation via DAHDI group 1
(Telenet))
exten => _00ZXXX,n,Dial(${TELENET}/${EXTEN:1})
exten => _00ZXXX,n,NoOp(DIALSTATUS is now ${DIALSTATUS})
exten => _00ZXXX,n,GoToIf($["${DIALSTATUS}" =
"ANSWER"]?free:occupied)
exten => _00ZXXX,n(free),NoOp(national conversation : dialstatus
free)
exten => _00ZXXX,n,NoOp(hangup-cause = ${HANGUPCAUSE})
exten => _00ZXXX,n,Hangup()
exten => _00ZXXX,n(occupied),NoOp(Telenetlijn occupied)
exten => _00ZXXX,n,Playtones(busy)
exten => _00ZXXX,n,Congestion(10)
exten => _00ZXXX,n,NoOp(hangup-cause = ${HANGUPCAUSE})
exten => _00ZXXX,n,Hangup()
exten => _00ZXXX,n,NoOp(hangup-oorzaak = ${HANGUPCAUSE})
I absolutely want to know how the call ended... but I don't seem to
capture the info I want.
So 2 questions :
1) if indeed the call ended abruptly, do you see something abnormal ?
2) How can I debug the way a call ended ? Be it in a normal way, or be
it by a sudden breakup.
Thank you very much,
Jonas.
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[asterisk-users] HANGUPCAUSE for unalocated number?
Hello, On your BRI or PRI's what do you guys get as HANGUPCAUSE when dialing an unalocated number? I always get 3 (no route) which is less than helpful. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] HANGUPCAUSE on SIP channels
Hopefully I'm not just missing some little detail here. We're trying to
set the HANGUPCAUSE on SIP channels to have our softswitch play the proper
recording instead of answering the call on Asterisk to play the message.
It appears that no matter what the HANGUPCAUSE is set to, Asterisk always
just sends "603 Declined".
I looked through the source code briefly and it appears that it *should*
work. It would be helpful to know if anyone actually uses this feature
and if it is working properly for them before we go through with fully
debugging and patching this to work for us.
Here is the our test extension from extensions.conf:
exten => 9218,1,Set(HANGUPCAUSE=1)
exten => 9218,2,Hangup
According to hangup_cause2sip in chan_sip.c a HANGUPCAUSE of 1 should
cause Asterisk to reply to the softswitch with a "404 Not Found" SIP
message. That doesn't seem to be the case, however.
Here is a bit of the verbose console output:
(Please note that I added some extra ast_log calls to the source code to
generate some extra debugging information.)
Apr 21 12:35:18 WARNING[16430]: pbx.c:5983 pbx_builtin_setvar_helper:
chan=SIP/nyct-901-539f, name=SIPURI, value=sip:[EMAIL PROTECTED]:5060
Apr 21 12:35:18 WARNING[16430]: pbx.c:5983 pbx_builtin_setvar_helper:
chan=SIP/nyct-901-539f, name=SIPDOMAIN, value=192.168.74.254
Apr 21 12:35:18 WARNING[16430]: pbx.c:5983 pbx_builtin_setvar_helper:
chan=SIP/nyct-901-539f, name=SIPUSERAGENT,
value=PolycomSoundPointIP-SPIP_501-UA/1.6.2.0041
Apr 21 12:35:18 WARNING[16430]: pbx.c:5983 pbx_builtin_setvar_helper:
chan=SIP/nyct-901-539f, name=SIPCALLID,
[EMAIL PROTECTED]
-- Executing Set("SIP/nyct-901-539f", "HANGUPCAUSE=1") in new stack
Apr 21 12:35:18 WARNING[16815]: pbx.c:5983 pbx_builtin_setvar_helper:
chan=SIP/nyct-901-539f, name=HANGUPCAUSE, value=1
Apr 21 12:35:18 WARNING[16815]: pbx.c:6057 pbx_builtin_setvar:
chan=SIP/nyct-901-539f, name=HANGUPCAUSE, value=1
-- Executing Hangup("SIP/nyct-901-539f", "") in new stack
Apr 21 12:35:18 WARNING[16815]: pbx.c:5548 pbx_builtin_hangup:
chan->hangupcause=(null)
== Spawn extension (nyct, 9218, 2) exited non-zero on
'SIP/nyct-901-539f'
Apr 21 12:35:18 WARNING[16815]: chan_sip.c:2471 sip_hangup:
ast->hangupcause=16 res=(null)
This is all on Asterisk 1.2.7.1. Your line numbers may vary since there
were some ast_log lines added. Hopefully this makes some sense to
someone.
Thanks for any help or input.
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Eric Futch <[EMAIL PROTECTED]> (212) 293-2620
Weather for KNYC: Apr 21 11:51a EDT, 59F (15C), Fair, Humidity 49%
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[Asterisk-Users] Hangupcause to handle Called party disconnect ? PSTN----E1----OldPBX---E1--Asterisk
Hi,I've been debuging the call disconnection problem in our
architecture:PSTN---E1---OldPBX---E1---AsteriskThis is our
problem:-SIP user agent "A" calls a pstn phone "B".-"B" hangs up the
call.-SIP user agent "A" starts listenning busytones... But the call still
on. (and being payed).- Call only ends when it is correctly hanged up in the
SIPphone.I've been tracing the communications between the OldPBX
(NETWORK) and Asterisk (USER SIDE) and i found this:M03 PROGRESS
I08 CauseCoding Std=CCITTLocation=Private net-remoteCause
Code=16I1E Progress indicatorCoding Std=CCITTLocation=Public
net-localProgress desc=Inband info availI28 DisplayInfo=CHAMADA
DESLIGADA08 02 00 02 03 08 0285 90 1E 02 82 88 2811 43 48 41 4D
41 4441 20 44 45 53 4C 4947 41 44 41RXB From User Side
00:45:29.902 Fr.25L2: Sapi=0 Tei=0INFOpf=0 Nr=84 Ns=6900 01
8A A8L3: PD=08 CR(D)=2M7D STATUS I08 CauseCoding
Std=CCITTLocation=UserCause Code=98I14 Call stateCoding
Std=CCITTState=1008 02 80 02 7D 08 0280 E2 14 01
0AThis trace reports to a called party that hanged up the call,
then our old PBX talked to Asterisk with :PROGRESSCause
Code=16and Asterisk answered with Location=UserCause
Code=98I've been looking ISDN cause Codes and i found:Cause No.
98 - message not compatible with call state or message type
non-existent.This cause indicates that the equipment sending this cause has
received a message such that the procedures do not indicate that this is a
permissible message to receive while in the call state, or a STATUS message was
received indicating an incompatible call state.I hope you can advice me.
Is it affordable to use Hangupcause?what we need is that, if the called
party hangs, asterisk should hang (safety reasons on billing)..exten
=> _2,1,Dial(Zap/g1/${EXTEN})exten =>
_2,2,gotoif,$[${HANGUPCAUSE} = 16]?9|1exten =>
9,1,HangupI'm not sure if this is possible neither recommended,
should be HangupCAUSE=16 or =98 ??Best regards,Marco Mouta
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Re: [Asterisk-Users] Hangupcause is not enough on PRI
Record the telco's message and play it as your own :P Pibix wrote: Hi, I’m using Asterisk and a TE110P E1 PRI in Chile. When I call to a disconnected number or any not operational number, the telco sends the Hangupcause disconnection code and an audio message notifying the disconnection cause to the user. Asterisk does not allow the user to hear the audio message form the telco, instead it cuts the call. Any other legacies PRI PBX I’ve tested allow the user to hear the audio message from the telco. A few months ago I was dealing with this problem (making the user hear the disconnection cause message from the telco) and someone suggested using the Hangupcause code (http://lists.digium.com/pipermail/asterisk-users/2005-December/133374.html), and it solved the problem momentarily. Now, when I call to a not operational number, depending on the Hangupcause variable, Asterisk plays an internal audio message notifying the user about the disconnection cause, but my client is not satisfied with that, he expect to hear the real audio messager form the telco. I would like to know if somebody solved this issue letting the user hear the real disconnection cause message form the telco. Thank you! Javier Ergas CEO Pibix.cl ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mojo <[EMAIL PROTECTED]> Office Manger, Horan & Company, LLC (907) 747- x112 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Hangupcause is not enough on PRI
Interesting about your tellco. A the tellco I have dealt with sent the DISCONNECT message when a non-operational number is called. The usual messages will come in the this order 1. proceeding 2. one or more progressing 3. disconnect with the cause value(if number is non-operational, or the network or destination is busy) On the other hand, upon receiving disconnect from tellco, Asterisk does not really need to disconnect right away if listening to the telco message is desired. (there might be a switch you can set for this). -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew Kohlsmith Sent: Tuesday, April 04, 2006 9:08 AM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Hangupcause is not enough on PRI On Tuesday 04 April 2006 08:47, Pibix wrote: > Asterisk does not allow the user to hear the audio message form the > telco, instead it cuts the call. Any other legacies PRI PBX I've > tested allow the user to hear the audio message from the telco. Can we see a pri debug trace for a call that does this? I think this might be more of a signaling issue, as my T1 PRI (Bell Canada) does not do this at all. Asterisk doesn't disconnect the call until it encounters something that issues a Hangup(), or the telco sends the DISCONNECT message. -A. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Hangupcause is not enough on PRI
On Tuesday 04 April 2006 08:47, Pibix wrote: > Asterisk does not allow the user to hear the audio message form the telco, > instead it cuts the call. Any other legacies PRI PBX I've tested allow the > user to hear the audio message from the telco. Can we see a pri debug trace for a call that does this? I think this might be more of a signaling issue, as my T1 PRI (Bell Canada) does not do this at all. Asterisk doesn't disconnect the call until it encounters something that issues a Hangup(), or the telco sends the DISCONNECT message. -A. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Hangupcause is not enough on PRI
Do not send me any more Best Regards, Mr.Peeramate Rochanasmita Project Manager/General Manager SIPphone (Thailand) Co., Ltd. 644/19 Moo 1 Klong Kum, Bung Kum Bangkok Thailand 10230 SIP No.100888 SIP Call Center No.888 Tel. +66 2690 3999 Fax. +66 2690 3535 Mobile. +66 1423 1423 Email : [EMAIL PROTECTED] MSN : [EMAIL PROTECTED] Website : www.sipphone.co.th From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Pibix Sent: Tuesday, April 04, 2006 7:48 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Hangupcause is not enough on PRI Hi, I’m using Asterisk and a TE110P E1 PRI in Chile. When I call to a disconnected number or any not operational number, the telco sends the Hangupcause disconnection code and an audio message notifying the disconnection cause to the user. Asterisk does not allow the user to hear the audio message form the telco, instead it cuts the call. Any other legacies PRI PBX I’ve tested allow the user to hear the audio message from the telco. A few months ago I was dealing with this problem (making the user hear the disconnection cause message from the telco) and someone suggested using the Hangupcause code (http://lists.digium.com/pipermail/asterisk-users/2005-December/133374.html), and it solved the problem momentarily. Now, when I call to a not operational number, depending on the Hangupcause variable, Asterisk plays an internal audio message notifying the user about the disconnection cause, but my client is not satisfied with that, he expect to hear the real audio messager form the telco. I would like to know if somebody solved this issue letting the user hear the real disconnection cause message form the telco. Thank you! Javier Ergas CEO Pibix.cl ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Hangupcause is not enough on PRI
Hi, I’m using Asterisk and a TE110P E1 PRI in Chile. When I call to a disconnected number or any not operational number, the telco sends the Hangupcause disconnection code and an audio message notifying the disconnection cause to the user. Asterisk does not allow the user to hear the audio message form the telco, instead it cuts the call. Any other legacies PRI PBX I’ve tested allow the user to hear the audio message from the telco. A few months ago I was dealing with this problem (making the user hear the disconnection cause message from the telco) and someone suggested using the Hangupcause code (http://lists.digium.com/pipermail/asterisk-users/2005-December/133374.html), and it solved the problem momentarily. Now, when I call to a not operational number, depending on the Hangupcause variable, Asterisk plays an internal audio message notifying the user about the disconnection cause, but my client is not satisfied with that, he expect to hear the real audio messager form the telco. I would like to know if somebody solved this issue letting the user hear the real disconnection cause message form the telco. Thank you! Javier Ergas CEO Pibix.cl ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] HANGUPCAUSE macro..
*I* consider it a bug. Mark (if I recall correctly) considers it "just the way it works". Matt Schulte wrote: Interesting, would this be considered a bug or is it rather intentional? Or is that a dumb question ;-) -Original Message- From: Eric Wieling [mailto:[EMAIL PROTECTED] IAX does not correctly set the HANGUPCAUSE for a LOT of things. Look at DIALSTATUS or look at the dial-result macro on http://www.fnords.org/~eric/asterisk/downloads/macros.inc begin:vcard fn:Eric Wileing n:Wileing;Eric email;internet:[EMAIL PROTECTED] tel;work:504-899-1387 x2120 x-mozilla-html:FALSE version:2.1 end:vcard ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] HANGUPCAUSE macro..
Interesting, would this be considered a bug or is it rather intentional? Or is that a dumb question ;-) -Original Message- From: Eric Wieling [mailto:[EMAIL PROTECTED] IAX does not correctly set the HANGUPCAUSE for a LOT of things. Look at DIALSTATUS or look at the dial-result macro on http://www.fnords.org/~eric/asterisk/downloads/macros.inc ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] HANGUPCAUSE macro..
IAX does not correctly set the HANGUPCAUSE for a LOT of things. Look at
DIALSTATUS or look at the dial-result macro on
http://www.fnords.org/~eric/asterisk/downloads/macros.inc
Matt Schulte wrote:
I am connecting Asterisk to Asterisk to PSTN (Either by SIP or PRI) and
am having some issues dealing with busy signals. I have the HANGUPCAUSE
dial result macro in place to generate my hangup causes. I get a
hangupcause on my "gateway" machine with a code of 34, here's the code:
... -snip-
exten => hangupcause+34,1,Busy
It does in fact pass this on to the "IAD" asterisk machine as it saw the
following:
-- IAX2/x.x.x.x:4569/1 is busy
-- Hungup 'IAX2/x.x.x.x:4569/1'
== Everyone is busy/congested at this time
-- Executing Macro("Zap/2-1", "dial-result") in new stack
-- Executing NoOp("Zap/2-1", "HANGUPCAUSE is 0") in new stack
The last line doesn't make sense to me, if it knows it's a busy signal
why on earth would it use code 0? (code 0 is "NOTDEFINED") Is there
another
way to handle these calls?
This is in my extensions.conf, as an example:
exten => _.,1,Dial(IAX2/:[EMAIL PROTECTED]/${EXTEN})
exten => _.,2,Macro(dial-result)
On priority 2 I want to avoid using "Busy", makes sense?
begin:vcard
fn:Eric Wileing
n:Wileing;Eric
email;internet:[EMAIL PROTECTED]
tel;work:504-899-1387 x2120
x-mozilla-html:FALSE
version:2.1
end:vcard
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[Asterisk-Users] HANGUPCAUSE macro..
I am connecting Asterisk to Asterisk to PSTN (Either by SIP or PRI) and
am having some issues dealing with busy signals. I have the HANGUPCAUSE
dial result macro in place to generate my hangup causes. I get a
hangupcause on my "gateway" machine with a code of 34, here's the code:
... -snip-
exten => hangupcause+34,1,Busy
It does in fact pass this on to the "IAD" asterisk machine as it saw the
following:
-- IAX2/x.x.x.x:4569/1 is busy
-- Hungup 'IAX2/x.x.x.x:4569/1'
== Everyone is busy/congested at this time
-- Executing Macro("Zap/2-1", "dial-result") in new stack
-- Executing NoOp("Zap/2-1", "HANGUPCAUSE is 0") in new stack
The last line doesn't make sense to me, if it knows it's a busy signal
why on earth would it use code 0? (code 0 is "NOTDEFINED") Is there
another
way to handle these calls?
This is in my extensions.conf, as an example:
exten => _.,1,Dial(IAX2/:[EMAIL PROTECTED]/${EXTEN})
exten => _.,2,Macro(dial-result)
On priority 2 I want to avoid using "Busy", makes sense?
Matt
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Re: [Asterisk-Users] HANGUPCAUSE
Over the weekend the problem with HANGUPCAUSE was fixed. HANGUPCAUSE now contains the Asterisk cause code. README.variables has been updated. The problem with Dial not seeing a busy on PRI lines have been fixed. I have a version of app_dial.c from 0.7.1 that's been patched with these two fixes. The fixes are also in CVS. Yay! On Mon, 2004-02-02 at 03:57, Tais M. Hansen wrote: > -BEGIN PGP SIGNED MESSAGE- > Hash: SHA1 > > On Friday 30 January 2004 17:57, Eric Wieling wrote: > > Personally I would like AST_CAUSE to be the Asterisk cause code (which > > should be the same for all technologies), TECH_CAUSE (IAX2_CAUSE, > > SIP_CAUSE, PRI_CAUSE) would be interesting and useful to some people. > > I didn't know SIP or IAX actually defined causes for hangups. I any case one > of us should make this a feature request on bugs.digium.com. > > - -- > Regards, > Tais M. Hansen > ComX Networks > Tel: +45-70257474 > Fax: +45-70257374 > -BEGIN PGP SIGNATURE- > Version: GnuPG v1.2.3 (GNU/Linux) > > iD8DBQFAHh752TEAILET3McRAjMAAKCM2KFPS1tWrLeWX7CWCwvXV9qeYgCff/9J > /aHOFJgyQ62wzenJLkRFBw4= > =ar0B > -END PGP SIGNATURE- > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- Go to http://www.digium.com/index.php?menu=documentation and look at the "Unofficial Links" section. This section has links to a wide variety of 3rd party Asterisk related pages. My page is the "Asterisk Resource Pages". BTEL Consulting 504-899-1387 or 850-484-4545 or 877-677-9643 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] HANGUPCAUSE
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Friday 30 January 2004 17:57, Eric Wieling wrote: > Personally I would like AST_CAUSE to be the Asterisk cause code (which > should be the same for all technologies), TECH_CAUSE (IAX2_CAUSE, > SIP_CAUSE, PRI_CAUSE) would be interesting and useful to some people. I didn't know SIP or IAX actually defined causes for hangups. I any case one of us should make this a feature request on bugs.digium.com. - -- Regards, Tais M. Hansen ComX Networks Tel: +45-70257474 Fax: +45-70257374 -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.3 (GNU/Linux) iD8DBQFAHh752TEAILET3McRAjMAAKCM2KFPS1tWrLeWX7CWCwvXV9qeYgCff/9J /aHOFJgyQ62wzenJLkRFBw4= =ar0B -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] HANGUPCAUSE
It would to be good in any way... :) - Original Message - From: "Tais M. Hansen" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Friday, January 30, 2004 12:57 PM Subject: Re: [Asterisk-Users] HANGUPCAUSE -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Friday 30 January 2004 15:59, CW_ASN - Gus wrote: > Ok, but is not working as expected... we can't see clear ISUP causes. We > can't make different treatments or store other causes than busy (cause=17) > in cdr's . You could use my approach and combine it with the CDR userfield. Personally I would like a PRI_CAUSE variable to be set as well as HANGUPCAUSE. - -- Regards, Tais M. Hansen ComX Networks Tel: +45-70257474 Fax: +45-70257374 -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.3 (GNU/Linux) iD8DBQFAGn8E2TEAILET3McRAuk4AJ4ljoWNtJSg/aPUOuodWwiC/MA1aQCgg/EG 5B+arXbMx37BtKSFLez3KlI= =61o0 -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] HANGUPCAUSE
See Bug Number 890 on bugs.digium.com.
--Eric
> From: "Tais M. Hansen" <[EMAIL PROTECTED]>
> To: <[EMAIL PROTECTED]>
> Sent: Friday, January 30, 2004 9:20 AM
> Subject: Re: [Asterisk-Users] Echo worsens in 0.7.1
>
>
> -BEGIN PGP SIGNED MESSAGE-
> Hash: SHA1
>
> On Friday 30 January 2004 00:57, Eric Wieling wrote:
> > Is there any chance 0.7.2 will include a fix for PRI Cause Codes not
> > being translated into Asterisk Cause Codes and being passed back to
> > app_dial (as well as fixing the apparently never working ${HANGUPCAUSE}
> > variable)?
>
> HANGUPCAUSE is working fine here (cvs).
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Re: [Asterisk-Users] HANGUPCAUSE
Personally I would like AST_CAUSE to be the Asterisk cause code (which should be the same for all technologies), TECH_CAUSE (IAX2_CAUSE, SIP_CAUSE, PRI_CAUSE) would be interesting and useful to some people. On Fri, 2004-01-30 at 09:57, Tais M. Hansen wrote: > -BEGIN PGP SIGNED MESSAGE- > Hash: SHA1 > > On Friday 30 January 2004 15:59, CW_ASN - Gus wrote: > > Ok, but is not working as expected... we can't see clear ISUP causes. We > > can't make different treatments or store other causes than busy (cause=17) > > in cdr's . > > You could use my approach and combine it with the CDR userfield. Personally I > would like a PRI_CAUSE variable to be set as well as HANGUPCAUSE. > > - -- > Regards, > Tais M. Hansen > ComX Networks > Tel: +45-70257474 > Fax: +45-70257374 > -BEGIN PGP SIGNATURE- > Version: GnuPG v1.2.3 (GNU/Linux) > > iD8DBQFAGn8E2TEAILET3McRAuk4AJ4ljoWNtJSg/aPUOuodWwiC/MA1aQCgg/EG > 5B+arXbMx37BtKSFLez3KlI= > =61o0 > -END PGP SIGNATURE- > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- Go to http://www.digium.com/index.php?menu=documentation and look at the "Unofficial Links" section. This section has links to a wide variety of 3rd party Asterisk related pages. My page is the "Asterisk Resource Pages". BTEL Consulting 504-899-1387 or 850-484-4545 or 877-677-9643 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] HANGUPCAUSE
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Friday 30 January 2004 15:59, CW_ASN - Gus wrote: > Ok, but is not working as expected... we can't see clear ISUP causes. We > can't make different treatments or store other causes than busy (cause=17) > in cdr's . You could use my approach and combine it with the CDR userfield. Personally I would like a PRI_CAUSE variable to be set as well as HANGUPCAUSE. - -- Regards, Tais M. Hansen ComX Networks Tel: +45-70257474 Fax: +45-70257374 -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.3 (GNU/Linux) iD8DBQFAGn8E2TEAILET3McRAuk4AJ4ljoWNtJSg/aPUOuodWwiC/MA1aQCgg/EG 5B+arXbMx37BtKSFLez3KlI= =61o0 -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] HANGUPCAUSE
Ok, but is not working as expected... we can't see clear ISUP causes. We
can't make different treatments or store other causes than busy (cause=17)
in cdr's .
Regards,
Gus
- Original Message -
From: "Tais M. Hansen" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Friday, January 30, 2004 9:48 AM
Subject: Re: [Asterisk-Users] HANGUPCAUSE
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On Friday 30 January 2004 13:31, CW_ASN - Gus wrote:
>> HANGUPCAUSE is working fine here (cvs).
> How? Is written in CDR?
CDRs contain BUSY when busy and NO ANSWER on the rest.
extensions.conf:
[provider-out]
...
exten => _XX.,7,Dial(ZAP/g1/${calledid}|120|r)
exten => _XX.,8,Goto(provider-out-failed|c${HANGUPCAUSE}|1)
[provider-out-failed]
exten => c1,1,Hangup()
exten => c2,1,Busy()
exten => c3,1,Answer()
exten => c3,2,ResetCDR()
exten => c3,3,Playtones(info)
exten => c3,4,Wait(60)
exten => c3,5,Hangup()
exten => c4,1,Congestion()
- --
Regards,
Tais M. Hansen
ComX Networks
Tel: +45-70257474
Fax: +45-70257374
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Version: GnuPG v1.2.3 (GNU/Linux)
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3FroTgPgWQmBrqGwjwktmvc=
=yyxo
-END PGP SIGNATURE-
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Re: [Asterisk-Users] HANGUPCAUSE
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On Friday 30 January 2004 13:31, CW_ASN - Gus wrote:
>> HANGUPCAUSE is working fine here (cvs).
> How? Is written in CDR?
CDRs contain BUSY when busy and NO ANSWER on the rest.
extensions.conf:
[provider-out]
...
exten => _XX.,7,Dial(ZAP/g1/${calledid}|120|r)
exten => _XX.,8,Goto(provider-out-failed|c${HANGUPCAUSE}|1)
[provider-out-failed]
exten => c1,1,Hangup()
exten => c2,1,Busy()
exten => c3,1,Answer()
exten => c3,2,ResetCDR()
exten => c3,3,Playtones(info)
exten => c3,4,Wait(60)
exten => c3,5,Hangup()
exten => c4,1,Congestion()
- --
Regards,
Tais M. Hansen
ComX Networks
Tel: +45-70257474
Fax: +45-70257374
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Version: GnuPG v1.2.3 (GNU/Linux)
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3FroTgPgWQmBrqGwjwktmvc=
=yyxo
-END PGP SIGNATURE-
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Re: [Asterisk-Users] HANGUPCAUSE
How? Is written in CDR?
Regards,
Gus
- Original Message -
From: "Tais M. Hansen" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Friday, January 30, 2004 9:20 AM
Subject: Re: [Asterisk-Users] Echo worsens in 0.7.1
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On Friday 30 January 2004 00:57, Eric Wieling wrote:
> Is there any chance 0.7.2 will include a fix for PRI Cause Codes not
> being translated into Asterisk Cause Codes and being passed back to
> app_dial (as well as fixing the apparently never working ${HANGUPCAUSE}
> variable)?
HANGUPCAUSE is working fine here (cvs).
- --
Regards,
Tais M. Hansen
ComX Networks
Tel: +45-70257474
Fax: +45-70257374
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=PbAl
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