Re: [asterisk-users] Capturing call Reject/Decline events on a PRI line
There is no event for Asterisk to recognize. The PROGRESS message just says that there is an audio message available for the caller to listen to. Asterisk just passes the indication to the peer channel and opens the audio path. It is the caller who must recognize any audio message that their call has been dropped. Thanks for the explanation. Any suggestion on how to recognise that the call has been dropped? As far as ISDN is concerned, the call has not been answered yet so Asterisk must keep waiting. As far as the ISDN signaling is concerned, the call is still going. There is no signaling to indicate the call is not going to proceed any further. Richard -- _ -- 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] Capturing call Reject/Decline events on a PRI line
Richard, I tried calling the same number outside of Asterisk, by making direct calls from a landline telephone and a mobile phones. When the user rejected the call, the call was immediately cancelled. This implies that for whatever reason, the call reject signal is not available for asterisk to process, even though the network processes the event. We use a digium TE420 card( http://www.digium.com/en/products/digital/te420.php) to connect to the PRI. To debug the scenario, how do I do the following: a. Is there an ISDN signal for call reject/denied? I'm sure there is, for reasons explained above. b. Is my service provider passing along the call reject/denied signal to the PRI? c. If the signal is passed along to the PRI, why is the card not recognising the signal? Call Reject is a pretty common feature and is in common use everywhere. There must be a simple way to fix this. -- Thanks, Ishwar. On Mon, Aug 1, 2011 at 7:36 PM, Richard Mudgett rmudg...@digium.com wrote: There is no event for Asterisk to recognize. The PROGRESS message just says that there is an audio message available for the caller to listen to. Asterisk just passes the indication to the peer channel and opens the audio path. It is the caller who must recognize any audio message that their call has been dropped. Thanks for the explanation. Any suggestion on how to recognise that the call has been dropped? As far as ISDN is concerned, the call has not been answered yet so Asterisk must keep waiting. As far as the ISDN signaling is concerned, the call is still going. There is no signaling to indicate the call is not going to proceed any further. Richard -- _ -- 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 -- _ -- 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] Capturing call Reject/Decline events on a PRI line
Richard, Thanks for the explanation. You were right about the lack of signalling to indicate that the call has been rejected, One particular service provider, instead of signalling rightaway that the call has been rejected, gives a voice message saying 'The user is busy. Please call later.', and doesn't send any more signals till the default timeout. Total face-palm moment :) -- Regards, Ishwar. On Mon, Aug 1, 2011 at 7:36 PM, Richard Mudgett rmudg...@digium.com wrote: There is no event for Asterisk to recognize. The PROGRESS message just says that there is an audio message available for the caller to listen to. Asterisk just passes the indication to the peer channel and opens the audio path. It is the caller who must recognize any audio message that their call has been dropped. Thanks for the explanation. Any suggestion on how to recognise that the call has been dropped? As far as ISDN is concerned, the call has not been answered yet so Asterisk must keep waiting. As far as the ISDN signaling is concerned, the call is still going. There is no signaling to indicate the call is not going to proceed any further. Richard -- _ -- 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 -- _ -- 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] Capturing call Reject/Decline events on a PRI line
Hi Richard, There is no event for Asterisk to recognize. The PROGRESS message just says that there is an audio message available for the caller to listen to. Asterisk just passes the indication to the peer channel and opens the audio path. It is the caller who must recognize any audio message that their call has been dropped. Thanks for the explanation. Any suggestion on how to recognise that the call has been dropped? -- Thanks, Ishwar. As far as ISDN is concerned, the call has not been answered yet so Asterisk must keep waiting. Richard -- _ -- 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 -- _ -- 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] Capturing call Reject/Decline events on a PRI line
find the inline comment... On 07/29/2011 12:11 AM, Ishwar Sridharan wrote: The dialplan is very simple. When the call comes in, we hand the call over to adhearsion. This is how the dialplan looks: ;group 0 will be used for incoming calls EXOIN = DAHDI/g0 ;group 11 for outgoing EXOOUT = DAHDI/G11 ;This will be used by adhearsion EXOCID= [general] autofallthrough = yes ;really? clearglobalvars = no [frompstn] ;Send everything to adhearsion exten = _X.,1,Ringing exten = _X.,n,AGI(agi://127.0.0.1 http://127.0.0.1) exten = _X.,n,Hangup() ; Please try this. ; End dialplan The rest of the logic happens in adhearsion. -- Thanks, Ishwar. On Thu, Jul 28, 2011 at 6:33 PM, Nikhil d.nik...@cem-solutions.net mailto:d.nik...@cem-solutions.net wrote: Can you share the dialplan ,where SIP call is dialing... Thanks Nikhil On 07/28/2011 06:15 PM, Ishwar Sridharan wrote: Hello everybody, We have an asterisk 1.8.4.1 setup, connected to a PRI line. We're currently facing an issue where asterisk does not recognise the event when the called party declines/cuts the call. This happens specifically over calls on a PRI line. For calls over SIP, call decline event is captured properly. I wasn't able to find a solution on the asterisk-users mailing list archive. Any suggestions/help would be much appreiciated :) I can share the relevant parts of the configuration files, if needed. Here's an excerpt from asterisk logs for a SIP call. -- SIP/x- requested special control 16, passing it to SIP/x-0001 -- Started music on hold, class 'default', on SIP/x-0001 -- SIP/x- requested special control 20, passing it to SIP/x-0001 -- Got SIP response 603 Decline back from 127.0.0.1:5063 http://127.0.0.1:5063/ -- SIP/x-0001 is busy -- Stopped music on hold on SIP/x-0001 As you can see, on a SIP call, a call reject event is identified. For a call over the PRI, on the other hand, this event is not recognised. Here's an excerpt from asterisk log for a call over PRI. Call from to . -- Requested transfer capability: 0x10 - 3K1AUDIO -- Called G11/x -- Started music on hold, class 'default', on DAHDI/i1/y -- DAHDI/i1/x-18f8 is proceeding passing it to DAHDI/i1/y -- DAHDI/i1/x-18f8 is ringing # At this point in time, x rejects the call. The event that's logged in asterisk is the following: -- DAHDI/i1/x-18f8 is making progress passing it to DAHDI/i1/y # And the call times out after the default 30s. -- Nobody picked up in 3 ms Is there a reason why asterisk doesn't recognise the call decline, and does it need any configuration changes to enable this? Thanks for your help. -- Cheers, Ishwar. -- _ -- Bandwidth and Colocation Provided byhttp://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 -- _ -- 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 -- _ -- 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 -- _ -- 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] Capturing call Reject/Decline events on a PRI line
Try to Add h extensions in frompstn context and print ${HANGUPCAUSE} in that
you will receive in that ,
also read this for better implementation.
http://www.voip-info.org/wiki/view/Asterisk+variable+hangupcause
regards
Dhaval
On Fri, Jul 29, 2011 at 11:58 AM, Nikhil d.nik...@cem-solutions.net wrote:
**
find the inline comment...
On 07/29/2011 12:11 AM, Ishwar Sridharan wrote:
The dialplan is very simple. When the call comes in, we hand the call over
to adhearsion.
This is how the dialplan looks:
;group 0 will be used for incoming calls
EXOIN = DAHDI/g0
;group 11 for outgoing
EXOOUT = DAHDI/G11
;This will be used by adhearsion
EXOCID=
[general]
autofallthrough = yes ;really?
clearglobalvars = no
[frompstn]
;Send everything to adhearsion
exten = _X.,1,Ringing
exten = _X.,n,AGI(agi://127.0.0.1)
exten = _X.,n,Hangup() ; Please try this.
; End dialplan
The rest of the logic happens in adhearsion.
--
Thanks,
Ishwar.
On Thu, Jul 28, 2011 at 6:33 PM, Nikhil d.nik...@cem-solutions.netwrote:
Can you share the dialplan ,where SIP call is dialing...
Thanks
Nikhil
On 07/28/2011 06:15 PM, Ishwar Sridharan wrote:
Hello everybody,
We have an asterisk 1.8.4.1 setup, connected to a PRI line.
We're currently facing an issue where asterisk does not recognise the
event when the called party declines/cuts the call. This happens
specifically over calls on a PRI line. For calls over SIP, call decline
event is captured properly.
I wasn't able to find a solution on the asterisk-users mailing list
archive. Any suggestions/help would be much appreiciated :) I can share the
relevant parts of the configuration files, if needed.
Here's an excerpt from asterisk logs for a SIP call.
-- SIP/x- requested special control 16, passing it to
SIP/x-0001
-- Started music on hold, class 'default', on SIP/x-0001
-- SIP/x- requested special control 20, passing it to
SIP/x-0001
-- Got SIP response 603 Decline back from 127.0.0.1:5063
-- SIP/x-0001 is busy
-- Stopped music on hold on SIP/x-0001
As you can see, on a SIP call, a call reject event is identified.
For a call over the PRI, on the other hand, this event is not recognised.
Here's an excerpt from asterisk log for a call over PRI.
Call from to .
-- Requested transfer capability: 0x10 - 3K1AUDIO
-- Called G11/x
-- Started music on hold, class 'default', on DAHDI/i1/y
-- DAHDI/i1/x-18f8 is proceeding passing it to DAHDI/i1/y
-- DAHDI/i1/x-18f8 is ringing
# At this point in time, x rejects the call. The event that's logged
in asterisk is the following:
-- DAHDI/i1/x-18f8 is making progress passing it to DAHDI/i1/y
# And the call times out after the default 30s.
-- Nobody picked up in 3 ms
Is there a reason why asterisk doesn't recognise the call decline, and
does it need any configuration changes to enable this?
Thanks for your help.
--
Cheers,
Ishwar.
--
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Re: [asterisk-users] Capturing call Reject/Decline events on a PRI line
HI Eric, Nikhil, Thanks a lot for the responses. Bear with me a little as I'm very new to asterisk. I reproduced the problem using standard dialplan. The following are the configuration files: *chan_dahdi.conf* *[trunkgroups] [channels] language=en nationalprefix=+91 pridialplan=national ; or national or local? usecallerid=yes hidecallerid=no callwaiting=yes allow_call_waiting_calls=no usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=no ;Might have to play with this. callerid=asreceived facilityenable=yes priindication=outofband cidsignalling=dtmf ; most likely dtmf based on the India link below cidstart=polarity_IN #include dahdi-channels.conf* *extensions.conf:* *[frompstn] exten = x,1,Ringing exten = x,n,Dial(Dahdi/G11/y) exten = x,n,Hangup()* When a calls x, we dial y This is what I find in the logs: -- Accepting call from 'a' to 'x on channel 0/24, span 1 -- Executing [x@frompstn:1] Ringing(DAHDI/i1/a-136, ) in new stack -- Executing [x@frompstn:2] Dial(DAHDI/i1/aa-136, Dahdi/G11/yy) in new stack -- Requested transfer capability: 0x10 - 3K1AUDIO -- Called G11/yy -- DAHDI/i1/y-137 is proceeding passing it to DAHDI/i1/a-136 -- DAHDI/i1/y-137 is ringing # At this point, y rejected the call. Asterisk doesn't recognise this, and continues to dial for 30s(the default) before hanging up. -- DAHDI/i1/y-137 is making progress passing it to DAHDI/i1/a-136 -- Nobody picked up in 3 ms I'll try out pri intense debug during night time when the traffic on our servers is low, and update the list with the logs. In the mean time, is there anything missing in the configuration that rejected calls aren't detected? -- Thanks, Ishwar. On Fri, Jul 29, 2011 at 1:10 AM, Eric Wieling ewiel...@nyigc.com wrote: 1) You have to have channels configured for your PRI SOMEWHERE in the Asterisk DAHDI configs. 2) Can't troubleshoot when everything important is masked by an AGI script. Reproduce the problem using standard dialplan stuff. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Ishwar Sridharan Sent: Thursday, July 28, 2011 2:52 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Capturing call Reject/Decline events on a PRI line Hi Eric, There weren't any lines with PRI channel = in the chan_dahdi.conf However, I added the lines you'd mentioned, near the top of the file. Still, no difference in either the behaviour or the asterisk output. Please note that as soon as the call lands on asterisk, we pass the control over to adhearsion. Does that affect how events are handled in asterisk? -- Thanks, Ishwar. On Thu, Jul 28, 2011 at 6:37 PM, Eric Wieling ewiel...@nyigc.com wrote: -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk- users- boun...@lists.digium.com] On Behalf Of Nikhil Sent: Thursday, July 28, 2011 9:03 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Capturing call Reject/Decline events on a PRI line Can you share the dialplan ,where SIP call is dialing... Thanks Nikhil On 07/28/2011 06:15 PM, Ishwar Sridharan wrote: Hello everybody, We have an asterisk 1.8.4.1 setup, connected to a PRI line. We're currently facing an issue where asterisk does not recognise the event when the called party declines/cuts the call. This happens specifically over calls on a PRI line. For calls over SIP, call decline event is captured properly. I wasn't able to find a solution on the asterisk-users mailing list archive. Any suggestions/help would be much appreiciated :) I can share the relevant parts of the configuration files, if needed. Here's an excerpt from asterisk logs for a SIP call. -- SIP/x- requested special control 16, passing it to SIP/x-0001 -- Started music on hold, class 'default', on SIP/x- 0001 -- SIP/x- requested special control 20, passing it to SIP/x-0001 -- Got SIP response 603 Decline back from 127.0.0.1:5063 http://127.0.0.1:5063/ -- SIP/x-0001 is busy -- Stopped music on hold on SIP/x-0001 As you can see, on a SIP call, a call reject event is identified. For a call
Re: [asterisk-users] Capturing call Reject/Decline events on a PRI line
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Ishwar Sridharan Sent: Friday, July 29, 2011 9:57 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Capturing call Reject/Decline events on a PRI line -- Called G11/yy -- DAHDI/i1/y-137 is proceeding passing it to DAHDI/i1/a-136 -- DAHDI/i1/y-137 is ringing # At this point, y rejected the call. Asterisk doesn't recognise this, and continues to dial for 30s(the default) before hanging up. -- DAHDI/i1/y-137 is making progress passing it to DAHDI/i1/a-136 -- Nobody picked up in 3 ms Exactly *how* is y rejecting the call? What is y? An ISDN Phone? A POTS phone? A PSTN telephone number? I am assuming that y is a PSTN TN. It is starting to sound like that is not the case. -- _ -- 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] Capturing call Reject/Decline events on a PRI line
HI Eric, is a mobile number in India, and the call id rejected by ending the call from the mobile. BTW, why is the mail going to asterisk-users-bounces? -- Thanks, Ishwar. On Fri, Jul 29, 2011 at 9:34 PM, Eric Wieling ewiel...@nyigc.com wrote: -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Ishwar Sridharan Sent: Friday, July 29, 2011 9:57 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Capturing call Reject/Decline events on a PRI line -- Called G11/yy -- DAHDI/i1/y-137 is proceeding passing it to DAHDI/i1/a-136 -- DAHDI/i1/y-137 is ringing # At this point, y rejected the call. Asterisk doesn't recognise this, and continues to dial for 30s(the default) before hanging up. -- DAHDI/i1/y-137 is making progress passing it to DAHDI/i1/a-136 -- Nobody picked up in 3 ms Exactly *how* is y rejecting the call? What is y? An ISDN Phone? A POTS phone? A PSTN telephone number? I am assuming that y is a PSTN TN. It is starting to sound like that is not the case. -- _ -- 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 -- _ -- 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] Capturing call Reject/Decline events on a PRI line
Hello, We enable pri intense debug with the standard asterisk PRI dialplan, collected the logs and you can find the logs attached to the mail. After the call was made, the called party cut the call, and asterisk doesn't seem to recognise the event. I can't make much sense of the logs given my non-existent background in telephony. Would somebody here help me figure why the event wasn't captured? -- Thanks, Ishwar. On Fri, Jul 29, 2011 at 11:54 PM, Ishwar Sridharan ish...@exotel.in wrote: HI Eric, is a mobile number in India, and the call id rejected by ending the call from the mobile. BTW, why is the mail going to asterisk-users-bounces? -- Thanks, Ishwar. On Fri, Jul 29, 2011 at 9:34 PM, Eric Wieling ewiel...@nyigc.com wrote: -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Ishwar Sridharan Sent: Friday, July 29, 2011 9:57 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Capturing call Reject/Decline events on a PRI line -- Called G11/yy -- DAHDI/i1/y-137 is proceeding passing it to DAHDI/i1/a-136 -- DAHDI/i1/y-137 is ringing # At this point, y rejected the call. Asterisk doesn't recognise this, and continues to dial for 30s(the default) before hanging up. -- DAHDI/i1/y-137 is making progress passing it to DAHDI/i1/a-136 -- Nobody picked up in 3 ms Exactly *how* is y rejecting the call? What is y? An ISDN Phone? A POTS phone? A PSTN telephone number? I am assuming that y is a PSTN TN. It is starting to sound like that is not the case. -- _ -- 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 1 Done handling message for SAPI/TEI=0/0 Span: 1 Processing event: PRI_EVENT_RINGING -- DAHDI/i1/09880847047-22c is ringing 1 TEI: 0 State 7(Multi-frame established) 1 V(A)=96, V(S)=96, V(R)=34 1 K=7, RC=1, l3initiated=0, reject_except=0, ack_pend=0 1 T200_id=0, N200=3, T203_id=1 1 [ 02 01 44 c0 08 02 81 1a 03 1e 02 8a 81 1e 02 8a 88 ] 1 Informational frame: 1 SAPI: 00 C/R: 1 EA: 0 1 TEI: 000EA: 1 1 N(S): 034 0: 0 1 N(R): 096 P: 0 1 13 bytes of data 1 Protocol Discriminator: Q.931 (8) len=13 1 TEI=0 Call Ref: len= 2 (reference 282/0x11A) (Sent to originator) 1 Message Type: PROGRESS (3) 1 [1e 02 8a 81] 1 Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Network beyond the interworking point (10) 1Ext: 1 Progress Description: Call is not end-to-end ISDN; further call progress information may be available inband. (1) ] 1 [1e 02 8a 88] 1 Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Network beyond the interworking point (10) 1Ext: 1 Progress Description: Inband information or appropriate pattern now available. (8) ] 1 -- Got ACK for N(S)=96 to (but not including) N(S)=96 1 -- T200 requested to stop when not started 1 T203 requested to start without stopping first 1 -- Starting T203 timer 1 Received message for call 0xd6885d0 on 0x2c5c0530 TEI/SAPI 0/0, call-pri is 0x2c5c0530 TEI/SAPI 0/0 1 -- Processing IE 30 (cs0, Progress Indicator) 1 -- Processing IE 30 (cs0, Progress Indicator) 1 1 TEI: 0 State 7(Multi-frame established) 1 V(A)=96, V(S)=96, V(R)=35 1 K=7, RC=1, l3initiated=0, reject_except=0, ack_pend=0 1 T200_id=0, N200=3, T203_id=1 1 [ 02 01 01 46 ] 1 Supervisory frame: 1 SAPI: 00 C/R: 1 EA: 0 1 TEI: 000EA: 1 1 Zero: 0 S: 0 01: 1 [ RR (receive ready) ] 1 N(R): 035 P/F: 0 1 0 bytes of data 1 Done handling message for SAPI/TEI=0/0 Span: 1 Processing event: PRI_EVENT_PROGRESS -- DAHDI/i1/09880847047-22c is making progress passing it to DAHDI/i1/8088919888-22b 1 TEI: 0 State 7(Multi-frame established) 1 V(A)=96, V(S)=96, V(R)=35 1 K=7, RC=1, l3initiated=0, reject_except=0, ack_pend=0 1 T200_id=0, N200=3, T203_id=0 1 [ 00 01 01 47 ] 1 Supervisory frame: 1 SAPI: 00 C/R: 0 EA: 0 1 TEI: 000EA: 1 1 Zero: 0 S: 0 01: 1 [ RR (receive ready) ] 1 N(R): 035 P/F: 1 1 0 bytes of data 1 -- Starting T200 timer 1 1 TEI: 0 State 8(Timer recovery) 1 V(A)=96, V(S)=96, V(R)=35 1 K=7, RC=0, l3initiated=0, reject_except=0, ack_pend=0 1 T200_id=1, N200=3, T203_id=0 1 [ 02 01 01 c1 ] 1 Supervisory frame: 1 SAPI: 00 C/R: 1 EA: 0 1 TEI: 000EA: 1 1 Zero: 0 S: 0 01: 1 [ RR (receive ready) ] 1 N(R): 096 P/F: 1 1 0 bytes of data 1 1 TEI: 0 State 8(Timer recovery) 1 V(A)=96, V(S)=96, V
Re: [asterisk-users] Capturing call Reject/Decline events on a PRI line
We enable pri intense debug with the standard asterisk PRI dialplan, collected the logs and you can find the logs attached to the mail. After the call was made, the called party cut the call, and asterisk doesn't seem to recognise the event. I can't make much sense of the logs given my non-existent background in telephony. Would somebody here help me figure why the event wasn't captured? There is no event for Asterisk to recognize. The PROGRESS message just says that there is an audio message available for the caller to listen to. Asterisk just passes the indication to the peer channel and opens the audio path. It is the caller who must recognize any audio message that their call has been dropped. As far as ISDN is concerned, the call has not been answered yet so Asterisk must keep waiting. Richard -- _ -- 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
[asterisk-users] Capturing call Reject/Decline events on a PRI line
Hello everybody, We have an asterisk 1.8.4.1 setup, connected to a PRI line. We're currently facing an issue where asterisk does not recognise the event when the called party declines/cuts the call. This happens specifically over calls on a PRI line. For calls over SIP, call decline event is captured properly. I wasn't able to find a solution on the asterisk-users mailing list archive. Any suggestions/help would be much appreiciated :) I can share the relevant parts of the configuration files, if needed. Here's an excerpt from asterisk logs for a SIP call. -- SIP/x- requested special control 16, passing it to SIP/x-0001 -- Started music on hold, class 'default', on SIP/x-0001 -- SIP/x- requested special control 20, passing it to SIP/x-0001 -- Got SIP response 603 Decline back from 127.0.0.1:5063 -- SIP/x-0001 is busy -- Stopped music on hold on SIP/x-0001 As you can see, on a SIP call, a call reject event is identified. For a call over the PRI, on the other hand, this event is not recognised. Here's an excerpt from asterisk log for a call over PRI. Call from to . -- Requested transfer capability: 0x10 - 3K1AUDIO -- Called G11/x -- Started music on hold, class 'default', on DAHDI/i1/y -- DAHDI/i1/x-18f8 is proceeding passing it to DAHDI/i1/y -- DAHDI/i1/x-18f8 is ringing # At this point in time, x rejects the call. The event that's logged in asterisk is the following: -- DAHDI/i1/x-18f8 is making progress passing it to DAHDI/i1/y # And the call times out after the default 30s. -- Nobody picked up in 3 ms Is there a reason why asterisk doesn't recognise the call decline, and does it need any configuration changes to enable this? Thanks for your help. -- Cheers, Ishwar. -- _ -- 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] Capturing call Reject/Decline events on a PRI line
Can you share the dialplan ,where SIP call is dialing... Thanks Nikhil On 07/28/2011 06:15 PM, Ishwar Sridharan wrote: Hello everybody, We have an asterisk 1.8.4.1 setup, connected to a PRI line. We're currently facing an issue where asterisk does not recognise the event when the called party declines/cuts the call. This happens specifically over calls on a PRI line. For calls over SIP, call decline event is captured properly. I wasn't able to find a solution on the asterisk-users mailing list archive. Any suggestions/help would be much appreiciated :) I can share the relevant parts of the configuration files, if needed. Here's an excerpt from asterisk logs for a SIP call. -- SIP/x- requested special control 16, passing it to SIP/x-0001 -- Started music on hold, class 'default', on SIP/x-0001 -- SIP/x- requested special control 20, passing it to SIP/x-0001 -- Got SIP response 603 Decline back from 127.0.0.1:5063 http://127.0.0.1:5063/ -- SIP/x-0001 is busy -- Stopped music on hold on SIP/x-0001 As you can see, on a SIP call, a call reject event is identified. For a call over the PRI, on the other hand, this event is not recognised. Here's an excerpt from asterisk log for a call over PRI. Call from to . -- Requested transfer capability: 0x10 - 3K1AUDIO -- Called G11/x -- Started music on hold, class 'default', on DAHDI/i1/y -- DAHDI/i1/x-18f8 is proceeding passing it to DAHDI/i1/y -- DAHDI/i1/x-18f8 is ringing # At this point in time, x rejects the call. The event that's logged in asterisk is the following: -- DAHDI/i1/x-18f8 is making progress passing it to DAHDI/i1/y # And the call times out after the default 30s. -- Nobody picked up in 3 ms Is there a reason why asterisk doesn't recognise the call decline, and does it need any configuration changes to enable this? Thanks for your help. -- Cheers, Ishwar. -- _ -- 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 -- _ -- 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] Capturing call Reject/Decline events on a PRI line
On Thursday 28 Jul 2011, Ishwar Sridharan wrote:
Is there a reason why asterisk doesn't recognise the call decline, and
does it need any configuration changes to enable this?
What are you seeing for ${HANGUPCAUSE} when this happens ? (Put a line such
as
exten = y, n, NoOp(Hangup cause was ${HANGUPCAUSE})
in your extensions.conf after the Dial() statement.)
Note that with traditional phones, there is actually no way to decline a call
besides answering it and asking the caller politely to go away :)
--
AJS
Answers come *after* questions.
--
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Re: [asterisk-users] Capturing call Reject/Decline events on a PRI line
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Nikhil Sent: Thursday, July 28, 2011 9:03 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Capturing call Reject/Decline events on a PRI line Can you share the dialplan ,where SIP call is dialing... Thanks Nikhil On 07/28/2011 06:15 PM, Ishwar Sridharan wrote: Hello everybody, We have an asterisk 1.8.4.1 setup, connected to a PRI line. We're currently facing an issue where asterisk does not recognise the event when the called party declines/cuts the call. This happens specifically over calls on a PRI line. For calls over SIP, call decline event is captured properly. I wasn't able to find a solution on the asterisk-users mailing list archive. Any suggestions/help would be much appreiciated :) I can share the relevant parts of the configuration files, if needed. Here's an excerpt from asterisk logs for a SIP call. -- SIP/x- requested special control 16, passing it to SIP/x-0001 -- Started music on hold, class 'default', on SIP/x-0001 -- SIP/x- requested special control 20, passing it to SIP/x-0001 -- Got SIP response 603 Decline back from 127.0.0.1:5063 http://127.0.0.1:5063/ -- SIP/x-0001 is busy -- Stopped music on hold on SIP/x-0001 As you can see, on a SIP call, a call reject event is identified. For a call over the PRI, on the other hand, this event is not recognised. Here's an excerpt from asterisk log for a call over PRI. Call from to . -- Requested transfer capability: 0x10 - 3K1AUDIO -- Called G11/x -- Started music on hold, class 'default', on DAHDI/i1/y -- DAHDI/i1/x-18f8 is proceeding passing it to DAHDI/i1/y -- DAHDI/i1/x-18f8 is ringing # At this point in time, x rejects the call. The event that's logged in asterisk is the following: -- DAHDI/i1/x-18f8 is making progress passing it to DAHDI/i1/y # And the call times out after the default 30s. -- Nobody picked up in 3 ms Is there a reason why asterisk doesn't recognise the call decline, and does it need any configuration changes to enable this? Thanks for your help. Try adding the following before your PRI channel = lines in your chan_dahdi.conf. If you are using a GUI like FreePBX, you will have place the info where you need to for FreePBX. facilityenable=yes priindication=outofband -- _ -- 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] Capturing call Reject/Decline events on a PRI line
The dialplan is very simple. When the call comes in, we hand the call over to adhearsion. This is how the dialplan looks: ;group 0 will be used for incoming calls EXOIN = DAHDI/g0 ;group 11 for outgoing EXOOUT = DAHDI/G11 ;This will be used by adhearsion EXOCID= [general] autofallthrough = yes ;really? clearglobalvars = no [frompstn] ;Send everything to adhearsion exten = _X.,1,Ringing exten = _X.,n,AGI(agi://127.0.0.1) ; End dialplan The rest of the logic happens in adhearsion. -- Thanks, Ishwar. On Thu, Jul 28, 2011 at 6:33 PM, Nikhil d.nik...@cem-solutions.net wrote: ** Can you share the dialplan ,where SIP call is dialing... Thanks Nikhil On 07/28/2011 06:15 PM, Ishwar Sridharan wrote: Hello everybody, We have an asterisk 1.8.4.1 setup, connected to a PRI line. We're currently facing an issue where asterisk does not recognise the event when the called party declines/cuts the call. This happens specifically over calls on a PRI line. For calls over SIP, call decline event is captured properly. I wasn't able to find a solution on the asterisk-users mailing list archive. Any suggestions/help would be much appreiciated :) I can share the relevant parts of the configuration files, if needed. Here's an excerpt from asterisk logs for a SIP call. -- SIP/x- requested special control 16, passing it to SIP/x-0001 -- Started music on hold, class 'default', on SIP/x-0001 -- SIP/x- requested special control 20, passing it to SIP/x-0001 -- Got SIP response 603 Decline back from 127.0.0.1:5063 -- SIP/x-0001 is busy -- Stopped music on hold on SIP/x-0001 As you can see, on a SIP call, a call reject event is identified. For a call over the PRI, on the other hand, this event is not recognised. Here's an excerpt from asterisk log for a call over PRI. Call from to . -- Requested transfer capability: 0x10 - 3K1AUDIO -- Called G11/x -- Started music on hold, class 'default', on DAHDI/i1/y -- DAHDI/i1/x-18f8 is proceeding passing it to DAHDI/i1/y -- DAHDI/i1/x-18f8 is ringing # At this point in time, x rejects the call. The event that's logged in asterisk is the following: -- DAHDI/i1/x-18f8 is making progress passing it to DAHDI/i1/y # And the call times out after the default 30s. -- Nobody picked up in 3 ms Is there a reason why asterisk doesn't recognise the call decline, and does it need any configuration changes to enable this? Thanks for your help. -- Cheers, Ishwar. -- _ -- 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 -- _ -- 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 -- _ -- 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] Capturing call Reject/Decline events on a PRI line
Hi AJS,
Our dialplan doesn't have a Dial() statement as that's taken care of by
adhearsion.
However, I added exten = y, n, NoOp(Hangup cause was ${HANGUPCAUSE})
at the end of our context, restarted asterisk.
The log doesn't have anything new.
--
Thanks,
Ishwar.
On Thu, Jul 28, 2011 at 6:38 PM, A J Stiles
asterisk_l...@earthshod.co.ukwrote:
On Thursday 28 Jul 2011, Ishwar Sridharan wrote:
Is there a reason why asterisk doesn't recognise the call decline, and
does it need any configuration changes to enable this?
What are you seeing for ${HANGUPCAUSE} when this happens ? (Put a line
such
as
exten = y, n, NoOp(Hangup cause was ${HANGUPCAUSE})
in your extensions.conf after the Dial() statement.)
Note that with traditional phones, there is actually no way to decline a
call
besides answering it and asking the caller politely to go away :)
--
AJS
Answers come *after* questions.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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Re: [asterisk-users] Capturing call Reject/Decline events on a PRI line
Hi Eric, There weren't any lines with PRI channel = in the chan_dahdi.conf However, I added the lines you'd mentioned, near the top of the file. Still, no difference in either the behaviour or the asterisk output. Please note that as soon as the call lands on asterisk, we pass the control over to adhearsion. Does that affect how events are handled in asterisk? -- Thanks, Ishwar. On Thu, Jul 28, 2011 at 6:37 PM, Eric Wieling ewiel...@nyigc.com wrote: -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Nikhil Sent: Thursday, July 28, 2011 9:03 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Capturing call Reject/Decline events on a PRI line Can you share the dialplan ,where SIP call is dialing... Thanks Nikhil On 07/28/2011 06:15 PM, Ishwar Sridharan wrote: Hello everybody, We have an asterisk 1.8.4.1 setup, connected to a PRI line. We're currently facing an issue where asterisk does not recognise the event when the called party declines/cuts the call. This happens specifically over calls on a PRI line. For calls over SIP, call decline event is captured properly. I wasn't able to find a solution on the asterisk-users mailing list archive. Any suggestions/help would be much appreiciated :) I can share the relevant parts of the configuration files, if needed. Here's an excerpt from asterisk logs for a SIP call. -- SIP/x- requested special control 16, passing it to SIP/x-0001 -- Started music on hold, class 'default', on SIP/x-0001 -- SIP/x- requested special control 20, passing it to SIP/x-0001 -- Got SIP response 603 Decline back from 127.0.0.1:5063 http://127.0.0.1:5063/ -- SIP/x-0001 is busy -- Stopped music on hold on SIP/x-0001 As you can see, on a SIP call, a call reject event is identified. For a call over the PRI, on the other hand, this event is not recognised. Here's an excerpt from asterisk log for a call over PRI. Call from to . -- Requested transfer capability: 0x10 - 3K1AUDIO -- Called G11/x -- Started music on hold, class 'default', on DAHDI/i1/y -- DAHDI/i1/x-18f8 is proceeding passing it to DAHDI/i1/y -- DAHDI/i1/x-18f8 is ringing # At this point in time, x rejects the call. The event that's logged in asterisk is the following: -- DAHDI/i1/x-18f8 is making progress passing it to DAHDI/i1/y # And the call times out after the default 30s. -- Nobody picked up in 3 ms Is there a reason why asterisk doesn't recognise the call decline, and does it need any configuration changes to enable this? Thanks for your help. Try adding the following before your PRI channel = lines in your chan_dahdi.conf. If you are using a GUI like FreePBX, you will have place the info where you need to for FreePBX. facilityenable=yes priindication=outofband -- _ -- 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 -- _ -- 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] Capturing call Reject/Decline events on a PRI line
1) You have to have channels configured for your PRI SOMEWHERE in the Asterisk DAHDI configs. 2) Can't troubleshoot when everything important is masked by an AGI script. Reproduce the problem using standard dialplan stuff. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Ishwar Sridharan Sent: Thursday, July 28, 2011 2:52 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Capturing call Reject/Decline events on a PRI line Hi Eric, There weren't any lines with PRI channel = in the chan_dahdi.conf However, I added the lines you'd mentioned, near the top of the file. Still, no difference in either the behaviour or the asterisk output. Please note that as soon as the call lands on asterisk, we pass the control over to adhearsion. Does that affect how events are handled in asterisk? -- Thanks, Ishwar. On Thu, Jul 28, 2011 at 6:37 PM, Eric Wieling ewiel...@nyigc.com wrote: -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk- users- boun...@lists.digium.com] On Behalf Of Nikhil Sent: Thursday, July 28, 2011 9:03 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Capturing call Reject/Decline events on a PRI line Can you share the dialplan ,where SIP call is dialing... Thanks Nikhil On 07/28/2011 06:15 PM, Ishwar Sridharan wrote: Hello everybody, We have an asterisk 1.8.4.1 setup, connected to a PRI line. We're currently facing an issue where asterisk does not recognise the event when the called party declines/cuts the call. This happens specifically over calls on a PRI line. For calls over SIP, call decline event is captured properly. I wasn't able to find a solution on the asterisk-users mailing list archive. Any suggestions/help would be much appreiciated :) I can share the relevant parts of the configuration files, if needed. Here's an excerpt from asterisk logs for a SIP call. -- SIP/x- requested special control 16, passing it to SIP/x-0001 -- Started music on hold, class 'default', on SIP/x- 0001 -- SIP/x- requested special control 20, passing it to SIP/x-0001 -- Got SIP response 603 Decline back from 127.0.0.1:5063 http://127.0.0.1:5063/ -- SIP/x-0001 is busy -- Stopped music on hold on SIP/x-0001 As you can see, on a SIP call, a call reject event is identified. For a call over the PRI, on the other hand, this event is not recognised. Here's an excerpt from asterisk log for a call over PRI. Call from to . -- Requested transfer capability: 0x10 - 3K1AUDIO -- Called G11/x -- Started music on hold, class 'default', on DAHDI/i1/y -- DAHDI/i1/x-18f8 is proceeding passing it to DAHDI/i1/y -- DAHDI/i1/x-18f8 is ringing # At this point in time, x rejects the call. The event that's logged in asterisk is the following: -- DAHDI/i1/x-18f8 is making progress passing it to DAHDI/i1/y # And the call times out after the default 30s. -- Nobody picked up in 3 ms Is there a reason why asterisk doesn't recognise the call decline, and does it need any configuration changes to enable this? Thanks for your help. Try adding the following before your PRI channel = lines in your chan_dahdi.conf. If you are using a GUI like FreePBX, you will have place the info where you need to for FreePBX. facilityenable=yes priindication=outofband -- _ -- 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 -- _ -- 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
