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Summary:ASTERISK-20391: No ringback tone on any call from Asterisk to connected NEC PBX Extensions
Reporter:Hashim (hashim)Labels:
Date Opened:2012-09-07 14:35:30Date Closed:2012-09-11 09:30:12
Priority:MajorRegression?
Status:Closed/CompleteComponents:
Versions:1.8.15.1 Frequency of
Occurrence		Constant
Related
Issues:
Environment:CentOS 5.8 32 bit, asterisk 1.8.15.1 Libpri 1.4.12. Dahdi 2.6.1 , Digium Hardware TE220 without echo (PRI 1 (A) PSTN E1 and PRI 2 (B) NEC E1, DID based routing to NEC PBX)Attachments:( 0) pri_debug_2012_09_08.txt
( 1) pri_debug.txt
Description:Hi,

Asterisk PBX connected to NEC (Model Number not available)ISDN signalling used qsig/euroisdn.
While calling from Caller X ->  PRI A (E1 PSTN) to DID Route to PRI B (NEC) -> NEC extension rings but no ringback on  Caller X but can talk if extension picks up the call.

It's looks like same issue mentioned in ASTERISK-14397
but in this case I can talk if the NEC extension picks up, only issue is no ringback tone for the external callers.
Internal Xlite/Eyebeam has ringback tone but Hard extensions cannot hear ringback tone.

I have had used/changed most available parameters for PRI.

{noformat}
[Sep  6 19:45:18] VERBOSE[8910] sig_pri.c:     -- Accepting call from '048873370' to '13227' on channel 0/1, span 1
[Sep  6 19:45:18] VERBOSE[8983] pbx.c:     -- Executing [13227@from-g11:1] GotoIf("DAHDI/i1/048873370-1", "0?no") in new stack
[Sep  6 19:45:18] VERBOSE[8983] pbx.c:     -- Executing [13227@from-g11:2] GotoIf("DAHDI/i1/048873370-1", "0?no") in new stack
[Sep  6 19:45:18] VERBOSE[8983] pbx.c:     -- Executing [13227@from-g11:3] Set("DAHDI/i1/048873370-1", "DATETIME=2012-0906-1945-18-1947381078-GST") in new stack
[Sep  6 19:45:18] VERBOSE[8983] pbx.c:     -- Executing [13227@from-g11:4] Answer("DAHDI/i1/048873370-1", "") in new stack
[Sep  6 19:45:18] VERBOSE[8983] pbx.c:     -- Executing [13227@from-g11:5] Set("DAHDI/i1/048873370-1", "CHANNEL(transfercapability)=SPEECH") in new stack
[Sep  6 19:45:18] VERBOSE[8983] pbx.c:     -- Executing [13227@from-g11:6] Dial("DAHDI/i1/048873370-1", "DAHDI/g12/13227,rTt") in new stack
[Sep  6 19:45:18] DEBUG[8983] sig_pri.c: sig_pri_request 32
[Sep  6 19:45:18] DEBUG[8983] sig_pri.c: CALLER NAME:  NUM: 048873370  --> Caller Number
[Sep  6 19:45:18] VERBOSE[8983] sig_pri.c:     -- Requested transfer capability: 0x00 - SPEECH
[Sep  6 19:45:18] VERBOSE[8983] app_dial.c:     -- Called DAHDI/g12/13227  --> NEC Trunk
[Sep  6 19:45:18] WARNING[8983] app_dial.c: Invalid timeout specified: 'rTt'. Setting timeout to infinite
[Sep  6 19:45:18] VERBOSE[8983] app_dial.c:     -- DAHDI/i2/13227-1 is proceeding passing it to DAHDI/i1/048873370-1
[Sep  6 19:45:18] VERBOSE[8983] app_dial.c:     -- DAHDI/i2/13227-1 is ringing --> NEC Extension Ringing but No Ringback tone for 048873370 (Only Silence) but if call answers then AUDIO is clean.
[Sep  6 19:45:24] VERBOSE[8910] sig_pri.c:     -- Span 1: Channel 0/1 got hangup request, cause 16
[Sep  6 19:45:24] VERBOSE[8983] chan_dahdi.c:     -- Hungup 'DAHDI/i2/13227-1'
[Sep  6 19:45:24] VERBOSE[8983] pbx.c:   == Spawn extension (from-g11, 13227, 6) exited non-zero on 'DAHDI/i1/048873370-1'
[Sep  6 19:45:24] VERBOSE[8983] chan_dahdi.c:     -- Hungup 'DAHDI/i1/048873370-1'
{noformat}


I have attached the debug log, no clue in that.
So far

1) Answer() before passing to PRI B NEC : No Success
2) Dial parameters r : No Success
3) Background : No Success
4) Generate ringing (Playtones): No Success, But this time I could hear one small ring then silence
5) PRI Signalling qsiq/euroisdn : No Success
6) PRI AUDIO 3KAUDIO to SPEECH: No Success
7) Change Asterisk Versions one time 1.8.12.0 : No Success

I appreciate some advice.
Thanks


Comments:By: Hashim (hashim) 2012-09-07 14:48:20.595-0500

PRI SPAN Debug log
By: Richard Mudgett (rmudgett) 2012-09-07 19:40:52.986-0500

Please do not use "pri set debug 2 span x" use "pri set debug on span x" instead.  You rarely need to look at layer 2 information and the Q.921 receiver ready frames just bury the more interesting Q.931 messages.

I don't see anything in the pri debug.  The code generates a ringback tone as a result of your use of the Ringing application.

You are missing a comma in the Dial() call which would be why the r option did not work.
{noformat}
[Sep  6 19:45:18] WARNING[8983] app_dial.c: Invalid timeout specified: 'rTt'. Setting timeout to infinite
{noformat}

By: Hashim (hashim) 2012-09-08 00:56:35.812-0500

I noticed that. I fixed the error and tried but the result same.
I have attached the new log file as you requested.

Thanks

By: Hashim (hashim) 2012-09-08 00:57:02.156-0500

PRI Debug 08-09-2012
By: Hashim (hashim) 2012-09-09 13:25:36.267-0500

Yesterday, I tried SVN Asterisk branch 1.8 .
No success again.

Alternatively I tried Dial options r(ring) and m but cannot find any affect to that options while the call send to channels.


By: Rusty Newton (rnewton) 2012-09-10 15:20:39.369-0500

We looked through your pri_debug_2012_09_08.txt

We saw two reasons you wouldn't be hearing any ringback *for the first call* in the debug:

1)For the first call in that debug, you still are missing an 'r' in the Dial string.

"[Sep  8 09:52:25] WARNING[13686] app_dial.c: Invalid timeout specified: 'r'. Setting timeout to infinite"

Therefore Asterisk does not generate an ALERTING with inband signaling

2)The second reason for that first call is the ALERTING coming from your NEC system does not include inband signaling information (ringback).


*For the second call*, you call the Ringing application, and Asterisk does send an ALERTING out with inband information on the first call leg. Whatever device is receiving that should hear ringback. If it doesn't, then that would be an issue with that device and not Asterisk.

Let us know if this clears that debug up and we'll close the issue out.
By: Hashim (hashim) 2012-09-11 03:08:06.040-0500

Yes, you are true.
Two reasons are causing the issues.
NEC was not giving inband information (ringback).
Now the dial 'r(ring)' or 'm' is working.
Debug information flooded and couldn't notice the dial warnings.

My issues are solved. Thanks to all