[Asterisk-Users] not getting ringing/busy/answer feedback on my PRI

Larry Shields LJ.Shields at Verizon.net
Sun Aug 29 13:53:38 MST 2004


 
This is my PRI Debug info for those interested in this problem:
 
PMDBRIDGE*CLI>
< Protocol Discriminator: Q.931 (8)  len=39
< Call Ref: len= 2 (reference 115/0x73) (Originator)
< Message type: SETUP (5)
< [04 03 90 90 a2]
< Bearer Capability (len= 5) [ Ext: 1  Q.931 Std: 0  Info transfer
capability: 3.1kHz audio (16)
<                              Ext: 1  Trans mode/rate: 64kbps, circuit-mode
(16)
<                              Ext: 1  User information layer 1: u-Law (34)
< [18 03 a9 83 83]
< Channel ID (len= 5) [ Ext: 1  IntID: Implicit, PRI Spare: 0, Exclusive
Dchan: 0
<                        ChanSel: Reserved
<                       Ext: 1  Coding: 0   Number Specified   Channel Type:
3
<                       Ext: 1  Channel: 3 ]
< [1e 02 81 83]
< Progress Indicator (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard (0) 0:
0   Location: Private network serving the local user (1)
<                               Ext: 1  Progress Description: Calling
equipment is non-ISDN. (3) ]
< [6c 0b a1 39 37 32 33 31 35 38 35 34 31]
< Calling Number (len=13) [ Ext: 1  TON: National Number (2)  NPI:
ISDN/Telephony Numbering Plan (E.164/E.163) (1)
<                           Presentation: Presentation permitted, user
number not screened (0) '8541' ]
< [70 05 a1 32 36 38 38]
< Called Number (len= 7) [ Ext: 1  TON: National Number (2)  NPI:
ISDN/Telephony Numbering Plan (E.164/E.163) (1) '2688' ]
-- Making new call for cr 115
-- Processing Q.931 Call Setup
-- Processing IE 4 (cs0, Bearer Capability)
-- Processing IE 24 (cs0, Channel Identification)
-- Processing IE 30 (cs0, Progress Indicator)
-- Processing IE 108 (cs0, Calling Party Number)
-- Processing IE 112 (cs0, Called Party Number)
> Protocol Discriminator: Q.931 (8)  len=14
> Call Ref: len= 2 (reference 32883/0x8073) (Terminator)
> Message type: SETUP ACKNOWLEDGE (13)
> [18 03 a9 83 83]
> Channel ID (len= 5) [ Ext: 1  IntID: Implicit, PRI Spare: 0, Exclusive
Dchan: 0
>                        ChanSel: Reserved
>                       Ext: 1  Coding: 0   Number Specified   Channel Type:
3
>                       Ext: 1  Channel: 3 ]
> [1e 02 81 82]
> Progress Indicator (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard (0) 0:
0   Location: Private network serving the local user (1)
>                               Ext: 1  Progress Description: Called
equipment is non-ISDN. (2) ]
    -- Accepting call from '8541' to '2688' on channel 0/3, span 1
    -- Executing Wait("Zap/3-1", "2") in new stack
< Protocol Discriminator: Q.931 (8)  len=13
< Call Ref: len= 2 (reference 115/0x73) (Originator)
< Message type: STATUS (125)
< [08 03 80 e1 0d]
< Cause (len= 5) [ Ext: 1  Coding: CCITT (ITU) standard (0) 0: 0   Location:
User (0)
<                  Ext: 1  Cause: Message type nonexist. (97), class =
Protocol Error (6) ]
<              Cause data 1: 0d (13)
< [14 01 01]I>
< Call State (len= 3) [ Ext: 0  Coding: CCITT (ITU) standard (0) Call state:
Call Initiated (1)
-- Processing IE 8 (cs0, Cause)
-- Processing IE 20 (cs0, Call State)
    -- Executing Answer("Zap/3-1", "") in new stack
> Protocol Discriminator: Q.931 (8)  len=14
> Call Ref: len= 2 (reference 32883/0x8073) (Terminator)
> Message type: CONNECT (7)
> [18 03 a9 83 83]
> Channel ID (len= 5) [ Ext: 1  IntID: Implicit, PRI Spare: 0, Exclusive
Dchan: 0
>                        ChanSel: Reserved
>                       Ext: 1  Coding: 0   Number Specified   Channel Type:
3
>                       Ext: 1  Channel: 3 ]
> [1e 02 81 82]
> Progress Indicator (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard (0) 0:
0   Location: Private network serving the local user (1)
>                               Ext: 1  Progress Description: Called
equipment is non-ISDN. (2) ]
    -- Executing MeetMe("Zap/3-1", "|Mps") in new stack
    -- Playing 'conf-getconfno' (language 'en')
PMDBRIDGE*CLI> pri
debug    intense  no       show
PMDBRIDGE*CLI> pri no debug span 1
Disabled debugging on span 1
    -- Playing 'conf-getconfno' (language 'en')


  _____  

From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Larry Shields
Sent: Sunday, August 29, 2004 3:42 PM
To: asterisk-users at lists.digium.com
Subject: [Asterisk-Users] not getting ringing/busy/answer feedback on my PRI


 
I posted a problem earlier thinking it was due to a lack of sound card.
Several members stated that you do not need a sound card to play audio to a
PRI channel.  I did some further testing and discovered that there is a
problem with call progress tones or signaling on my PRI.  I think that the
reason I am not hearing audio from the MeetMe() or Playback() apps. is
because the the calling side of the PRI (NEC IPX), is not seeing the Answer
signal. I believe it is waiting for a ring and/or answer condition even
after Asterisk has executed an Answer() and Playback().  
 
The only other problem that I am having with my setup is that the
CONSOLE/DSP is not functional... I am not sure if the two problems are
related.  Any help is appreciated.  Please see my two examples below:
 
 
Unless my incoming DID (2000),  is pointed to a SIP station that is
registered and functional, I do not receive call progress tones on inbound
calls.
 
If I point the DID to an application like:
 
 [inbound_pri]
; PRI from the NEAX2400
 
exten => 2000,1,Wait,3
exten => 2000,2,Answer
exten => 2000,3,MeetMe,|Mps
exten => 2000,4,Hangup
 
I will not hear any initial ringback, and once answered there will be no
audio on the channel.
 
If I point the DID to a registered SIP station like:
 
[inbound_pri]
; PRI from the NEAX2400
 
exten => 2000,1,Wait,3
exten => 2000,2,Dial,SIP/2001,15,Tr
exten => 2000,Hangup
 
It will provide ringback tone to the calling channel on the PRI, and when
the ringing SIP phone answers there will be 2-way speech path.
 
 
 

  
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040829/6dc3a649/attachment.htm


More information about the asterisk-users mailing list