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<DIV><FONT face=Arial size=2><SPAN class=187305020-29082004>This is my PRI Debug
info for those interested in this problem:</SPAN></FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>PMDBRIDGE*CLI><BR>< Protocol Discriminator:
Q.931 (8) len=39<BR>< Call Ref: len= 2 (reference 115/0x73)
(Originator)<BR>< Message type: SETUP (5)<BR>< [04 03 90 90 a2]<BR><
Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer
capability: 3.1kHz audio
(16)<BR><
Ext: 1 Trans mode/rate: 64kbps, circuit-mode
(16)<BR><
Ext: 1 User information layer 1: u-Law (34)<BR>< [18 03 a9 83
83]<BR>< Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0,
Exclusive Dchan:
0<BR><
ChanSel:
Reserved<BR><
Ext: 1 Coding: 0 Number Specified Channel Type:
3<BR><
Ext: 1 Channel: 3 ]<BR>< [1e 02 81 83]<BR>< Progress Indicator (len=
4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location:
Private network serving the local user
(1)<BR><
Ext: 1 Progress Description: Calling equipment is non-ISDN. (3) ]<BR><
[6c 0b a1 39 37 32 33 31 35 38 35 34 31]<BR>< Calling Number (len=13) [ Ext:
1 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan
(E.164/E.163)
(1)<BR><
Presentation: Presentation permitted, user number not screened (0) '8541'
]<BR>< [70 05 a1 32 36 38 38]<BR>< Called Number (len= 7) [ Ext: 1
TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163)
(1) '2688' ]<BR>-- Making new call for cr 115<BR>-- Processing Q.931 Call
Setup<BR>-- Processing IE 4 (cs0, Bearer Capability)<BR>-- Processing IE 24
(cs0, Channel Identification)<BR>-- Processing IE 30 (cs0, Progress
Indicator)<BR>-- Processing IE 108 (cs0, Calling Party Number)<BR>-- Processing
IE 112 (cs0, Called Party Number)<BR>> Protocol Discriminator: Q.931
(8) len=14<BR>> Call Ref: len= 2 (reference 32883/0x8073)
(Terminator)<BR>> Message type: SETUP ACKNOWLEDGE (13)<BR>> [18 03 a9 83
83]<BR>> Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0,
Exclusive Dchan:
0<BR>>
ChanSel:
Reserved<BR>>
Ext: 1 Coding: 0 Number Specified Channel Type:
3<BR>>
Ext: 1 Channel: 3 ]<BR>> [1e 02 81 82]<BR>> Progress Indicator (len=
4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location:
Private network serving the local user
(1)<BR>>
Ext: 1 Progress Description: Called equipment is non-ISDN. (2)
]<BR> -- Accepting call from '8541' to '2688' on channel 0/3,
span 1<BR> -- Executing Wait("Zap/3-1", "2") in new
stack<BR>< Protocol Discriminator: Q.931 (8) len=13<BR>< Call Ref:
len= 2 (reference 115/0x73) (Originator)<BR>< Message type: STATUS
(125)<BR>< [08 03 80 e1 0d]<BR>< Cause (len= 5) [ Ext: 1 Coding:
CCITT (ITU) standard (0) 0: 0 Location: User
(0)<BR><
Ext: 1 Cause: Message type nonexist. (97), class = Protocol Error (6)
]<BR><
Cause data 1: 0d (13)<BR>< [14 01 01]I><BR>< Call State (len= 3) [ Ext:
0 Coding: CCITT (ITU) standard (0) Call state: Call Initiated (1)<BR>--
Processing IE 8 (cs0, Cause)<BR>-- Processing IE 20 (cs0, Call
State)<BR> -- Executing Answer("Zap/3-1", "") in new
stack<BR>> Protocol Discriminator: Q.931 (8) len=14<BR>> Call Ref:
len= 2 (reference 32883/0x8073) (Terminator)<BR>> Message type: CONNECT
(7)<BR>> [18 03 a9 83 83]<BR>> Channel ID (len= 5) [ Ext: 1 IntID:
Implicit, PRI Spare: 0, Exclusive Dchan:
0<BR>>
ChanSel:
Reserved<BR>>
Ext: 1 Coding: 0 Number Specified Channel Type:
3<BR>>
Ext: 1 Channel: 3 ]<BR>> [1e 02 81 82]<BR>> Progress Indicator (len=
4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location:
Private network serving the local user
(1)<BR>>
Ext: 1 Progress Description: Called equipment is non-ISDN. (2)
]<BR> -- Executing MeetMe("Zap/3-1", "|Mps") in new
stack<BR> -- Playing 'conf-getconfno' (language
'en')<BR>PMDBRIDGE*CLI> pri<BR>debug intense
no show<BR>PMDBRIDGE*CLI> pri no debug
span 1<BR>Disabled debugging on span 1<BR> -- Playing
'conf-getconfno' (language 'en')<BR></DIV></FONT></DIV><BR>
<DIV class=OutlookMessageHeader lang=en-us dir=ltr align=left>
<HR tabIndex=-1>
<FONT face=Tahoma size=2><B>From:</B> asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] <B>On Behalf Of </B>Larry
Shields<BR><B>Sent:</B> Sunday, August 29, 2004 3:42 PM<BR><B>To:</B>
asterisk-users@lists.digium.com<BR><B>Subject:</B> [Asterisk-Users] not getting
ringing/busy/answer feedback on my PRI<BR></FONT><BR></DIV>
<DIV></DIV>
<DIV><FONT face=Arial size=2><SPAN
class=546301419-29082004></SPAN></FONT> </DIV>
<DIV><FONT face=Arial size=2><SPAN class=546301419-29082004>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().
<DIV><FONT face=Arial size=2><SPAN
class=546301419-29082004></SPAN></FONT> </DIV>
<DIV><FONT face=Arial size=2><SPAN class=546301419-29082004>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:</SPAN></FONT></DIV></SPAN></FONT></DIV>
<DIV><FONT face=Arial size=2><SPAN class=546301419-29082004></SPAN></FONT><FONT
face=Arial size=2><SPAN class=546301419-29082004></SPAN></FONT> </DIV>
<DIV><FONT face=Arial size=2><SPAN
class=546301419-29082004></SPAN></FONT> </DIV>
<DIV><FONT face=Arial size=2><SPAN class=546301419-29082004>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.</SPAN></FONT></DIV>
<DIV><FONT face=Arial size=2><SPAN
class=546301419-29082004></SPAN></FONT> </DIV>
<DIV><FONT face=Arial size=2><SPAN class=546301419-29082004>If I point the DID
to an application like:</SPAN></FONT></DIV>
<DIV><FONT face=Arial size=2><SPAN
class=546301419-29082004></SPAN></FONT> </DIV>
<DIV><FONT face=Arial size=2><SPAN
class=546301419-29082004> [inbound_pri]<BR>; PRI from the
NEAX2400</SPAN></FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2><SPAN class=546301419-29082004>exten =>
2000,1,Wait,3<BR>exten => 2000,2,Answer<BR>exten =>
2000,3,MeetMe,|Mps<BR>exten => 2000,4,Hangup</SPAN></FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2><SPAN class=546301419-29082004>I will not hear any
initial ringback, and once answered there will be no audio on the
channel.</SPAN></FONT></DIV>
<DIV><FONT face=Arial size=2><SPAN
class=546301419-29082004></SPAN></FONT> </DIV>
<DIV><FONT face=Arial size=2><SPAN class=546301419-29082004>If I point the DID
to a registered SIP station like:</SPAN></FONT></DIV>
<DIV><FONT face=Arial size=2><SPAN
class=546301419-29082004></SPAN></FONT> </DIV>
<DIV><FONT face=Arial size=2><SPAN class=546301419-29082004>[inbound_pri]<BR>;
PRI from the NEAX2400</SPAN></FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2><SPAN class=546301419-29082004>exten =>
2000,1,Wait,3<BR>exten => 2000,2,Dial,SIP/2001,15,Tr<BR>exten =>
2000,Hangup</SPAN></FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2><SPAN class=546301419-29082004>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.</SPAN></FONT></DIV>
<DIV><FONT face=Arial size=2><SPAN
class=546301419-29082004></SPAN></FONT> </DIV>
<DIV><FONT face=Arial size=2><SPAN
class=546301419-29082004></SPAN></FONT> </DIV>
<DIV><FONT face=Arial size=2><SPAN class=546301419-29082004> </DIV>
<DIV><BR> </DIV></SPAN></FONT></BODY></HTML>