Larry Shields
2004-Aug-29 13:41 UTC
[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/20040829/8919f93e/attachment.htm
Larry Shields
2004-Aug-29 13:53 UTC
[Asterisk-Users] not getting ringing/busy/answer feedback on my PRI
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, ExclusiveDchan: 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: Calledequipment 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, ExclusiveDchan: 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: Calledequipment 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@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Larry Shields Sent: Sunday, August 29, 2004 3:42 PM To: asterisk-users@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