Dave George
2012-Feb-25 01:06 UTC
[asterisk-users] No IVR audio. Jump in RTP sequence number
My users dial *120 get to an IVR menu that plays their balance and then
ask them for a voucher. Ater the balance is played and the request for
the voucher is played the user don't hear any other audio from the
asterisk box. I can see the asterisk server playing the files to ask
for the voucher again but the user cannot hear any thing.
Has any one seens this issue with IVRs. I notice a change in RTP
sequence when voucher is being requested again.
sip debug
<--- SIP read from UDP:x.x.x.x:5060 --->
INVITE sip:*120 at a.b.c.d SIP/2.0
Via: SIP/2.0/UDP x.x.x.x;rport;branch=z9hG4bK2vr8B7r60vS7j
Max-Forwards: 70
From: "14735201326" <sip:14735201326 at
x.x.x.x>;tag=0K219XHeF7K2j
To: <sip:*120 at a.b.c.d>
Call-ID: 00ddbda6-d9b1-122f-e7a7-00259025b560
CSeq: 24716447 INVITE
Contact: <sip:14735201326 at x.x.x.x:5060>
User-Agent: Wireless Call Manager
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
Supported: timer, precondition, path, replaces
Allow-Events: talk, presence, dialog, call-info, sla,
include-session-description, presence.winfo, message-summary, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 231
Remote-Party-ID: "14735201326"
<sip:14735201326 at x.x.x.x>;party=calling;screen=yes;privacy=off
v=0
o=wCM 1330087502 1330087503 IN IP4 x.x.x.x
s=wCM
c=IN IP4 x.x.x.x
t=0 0
m=audio 17520 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
<------------->
--- (16 headers 11 lines) ---
Sending to x.x.x.x:5060 (no NAT)
Using INVITE request as basis request -
00ddbda6-d9b1-122f-e7a7-00259025b560
Found peer 'STARMG1' for '14735201326' from x.x.x.x:5060
== Using SIP RTP CoS mark 5
Found RTP audio format 18
Found RTP audio format 101
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - 0x100 (g729), peer - audio=0x100 (g729)/video=0x0
(nothing)/text=0x0 (nothing), combined - 0x100 (g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1
(telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port x.x.x.x:17520
Looking for *120 in spicemobile (domain a.b.c.d)
list_route: hop: <sip:14735201326 at x.x.x.x:5060>
<--- Transmitting (no NAT) to x.x.x.x:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
x.x.x.x;branch=z9hG4bK2vr8B7r60vS7j;received=x.x.x.x;rport=5060
From: "14735201326" <sip:14735201326 at
x.x.x.x>;tag=0K219XHeF7K2j
To: <sip:*120 at a.b.c.d>
Call-ID: 00ddbda6-d9b1-122f-e7a7-00259025b560
CSeq: 24716447 INVITE
Server: Asterisk PBX 1.8.7.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:*120 at a.b.c.d:5060>
Content-Length: 0
<------------>
<--- SIP read from UDP:x.x.x.x:5060 --->
INVITE sip:*120 at a.b.c.d SIP/2.0
Via: SIP/2.0/UDP x.x.x.x;rport;branch=z9hG4bK2vr8B7r60vS7j
Max-Forwards: 70
From: "14735201326" <sip:14735201326 at
x.x.x.x>;tag=0K219XHeF7K2j
To: <sip:*120 at a.b.c.d>
Call-ID: 00ddbda6-d9b1-122f-e7a7-00259025b560
CSeq: 24716447 INVITE
Contact: <sip:14735201326 at x.x.x.x:5060>
User-Agent: Wireless Call Manager
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
Supported: timer, precondition, path, replaces
Allow-Events: talk, presence, dialog, call-info, sla,
include-session-description, presence.winfo, message-summary, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 231
Remote-Party-ID: "14735201326"
<sip:14735201326 at x.x.x.x>;party=calling;screen=yes;privacy=off
v=0
o=wCM 1330087502 1330087503 IN IP4 x.x.x.x
s=wCM
c=IN IP4 x.x.x.x
t=0 0
m=audio 17520 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
<------------->
--- (16 headers 11 lines) ---
Ignoring this INVITE request
<--- Transmitting (no NAT) to x.x.x.x:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
x.x.x.x;branch=z9hG4bK2vr8B7r60vS7j;received=x.x.x.x;rport=5060
From: "14735201326" <sip:14735201326 at
x.x.x.x>;tag=0K219XHeF7K2j
To: <sip:*120 at a.b.c.d>
Call-ID: 00ddbda6-d9b1-122f-e7a7-00259025b560
CSeq: 24716447 INVITE
Server: Asterisk PBX 1.8.7.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:*120 at a.b.c.d:5060>
Content-Length: 0
<------------>
-- Executing [*120 at spicemobile:1] AGI("SIP/STARMG1-000003c0",
"a2billing.php,6,voucher") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/a2billing.php
<SIP/STARMG1-000003c0>AGI Tx >> agi_request: a2billing.php
<SIP/STARMG1-000003c0>AGI Tx >> agi_channel: SIP/STARMG1-000003c0
<SIP/STARMG1-000003c0>AGI Tx >> agi_language: en
<SIP/STARMG1-000003c0>AGI Tx >> agi_type: SIP
<SIP/STARMG1-000003c0>AGI Tx >> agi_uniqueid: 1330130582.960
<SIP/STARMG1-000003c0>AGI Tx >> agi_version: 1.8.7.1
<SIP/STARMG1-000003c0>AGI Tx >> agi_callerid: 14735201326
<SIP/STARMG1-000003c0>AGI Tx >> agi_calleridname: 14735201326
<SIP/STARMG1-000003c0>AGI Tx >> agi_callingpres: 0
<SIP/STARMG1-000003c0>AGI Tx >> agi_callingani2: 0
<SIP/STARMG1-000003c0>AGI Tx >> agi_callington: 0
<SIP/STARMG1-000003c0>AGI Tx >> agi_callingtns: 0
<SIP/STARMG1-000003c0>AGI Tx >> agi_dnid: *120
<SIP/STARMG1-000003c0>AGI Tx >> agi_rdnis: unknown
<SIP/STARMG1-000003c0>AGI Tx >> agi_context: spicemobile
<SIP/STARMG1-000003c0>AGI Tx >> agi_extension: *120
<SIP/STARMG1-000003c0>AGI Tx >> agi_priority: 1
<SIP/STARMG1-000003c0>AGI Tx >> agi_enhanced: 0.0
<SIP/STARMG1-000003c0>AGI Tx >> agi_accountcode:
<SIP/STARMG1-000003c0>AGI Tx >> agi_threadid: 1118284096
<SIP/STARMG1-000003c0>AGI Tx >> agi_arg_1: 6
<SIP/STARMG1-000003c0>AGI Tx >> agi_arg_2: voucher
<SIP/STARMG1-000003c0>AGI Tx >>
<SIP/STARMG1-000003c0>AGI Rx << GET VARIABLE IDCONF
<SIP/STARMG1-000003c0>AGI Tx >> 200 result=0
<SIP/STARMG1-000003c0>AGI Rx << ANSWER
Audio is at 5060
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<--- Reliably Transmitting (no NAT) to x.x.x.x:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP
x.x.x.x;branch=z9hG4bK2vr8B7r60vS7j;received=x.x.x.x;rport=5060
From: "14735201326" <sip:14735201326 at
x.x.x.x>;tag=0K219XHeF7K2j
To: <sip:*120 at a.b.c.d>;tag=as20a616d1
Call-ID: 00ddbda6-d9b1-122f-e7a7-00259025b560
CSeq: 24716447 INVITE
Server: Asterisk PBX 1.8.7.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:*120 at a.b.c.d:5060>
Content-Type: application/sdp
Content-Length: 286
v=0
o=root 428800944 428800944 IN IP4 a.b.c.d
s=Asterisk PBX 1.8.7.1
c=IN IP4 a.b.c.d
t=0 0
m=audio 19238 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<------------>
<--- SIP read from UDP:x.x.x.x:5060 --->
ACK sip:*120 at a.b.c.d:5060 SIP/2.0
Via: SIP/2.0/UDP x.x.x.x;rport;branch=z9hG4bK35H1D299X5Fte
Max-Forwards: 70
From: "14735201326" <sip:14735201326 at
x.x.x.x>;tag=0K219XHeF7K2j
To: <sip:*120 at a.b.c.d>;tag=as20a616d1
Call-ID: 00ddbda6-d9b1-122f-e7a7-00259025b560
CSeq: 24716447 ACK
Contact: <sip:14735201326 at x.x.x.x:5060>
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
<SIP/STARMG1-000003c0>AGI Tx >> 200 result=0
<SIP/STARMG1-000003c0>AGI Rx << SET VARIABLE CHANNEL(language)
"en"
<SIP/STARMG1-000003c0>AGI Tx >> 200 result=1
<SIP/STARMG1-000003c0>AGI Rx << STREAM FILE prepaid-you-have
"#" 0
-- Playing 'prepaid-you-have' (escape_digits=#) (sample_offset 0)
<SIP/STARMG1-000003c0>AGI Tx >> 200 result=0 endpos=9440
<SIP/STARMG1-000003c0>AGI Rx << SAY NUMBER 52 ""
-- <SIP/STARMG1-000003c0> Playing 'digits/50.gsm' (language
'en')
-- <SIP/STARMG1-000003c0> Playing 'digits/2.gsm' (language
'en')
<SIP/STARMG1-000003c0>AGI Tx >> 200 result=0
<SIP/STARMG1-000003c0>AGI Rx << STREAM FILE dollars "#" 0
-- Playing 'dollars' (escape_digits=#) (sample_offset 0)
<SIP/STARMG1-000003c0>AGI Tx >> 200 result=0 endpos=7200
<SIP/STARMG1-000003c0>AGI Rx << STREAM FILE vm-and "#" 0
-- Playing 'vm-and' (escape_digits=#) (sample_offset 0)
<SIP/STARMG1-000003c0>AGI Tx >> 200 result=0 endpos=4640
<SIP/STARMG1-000003c0>AGI Rx << SAY NUMBER 97 ""
-- <SIP/STARMG1-000003c0> Playing 'digits/90.gsm' (language
'en')
-- <SIP/STARMG1-000003c0> Playing 'digits/7.gsm' (language
'en')
<SIP/STARMG1-000003c0>AGI Tx >> 200 result=0
<SIP/STARMG1-000003c0>AGI Rx << STREAM FILE prepaid-cents
"#" 0
-- Playing 'prepaid-cents' (escape_digits=#) (sample_offset 0)
<SIP/STARMG1-000003c0>AGI Tx >> 200 result=0 endpos=5600
<SIP/STARMG1-000003c0>AGI Rx << GET DATA
prepaid-voucher_enter_number
20000 9 #
-- <SIP/STARMG1-000003c0> Playing
'prepaid-voucher_enter_number.gsm'
(language 'en')
<SIP/STARMG1-000003c0>AGI Tx >> 200 result=326452854
no audio from here
<SIP/STARMG1-000003c0>AGI Rx << STREAM FILE voucher_does_not_exist
"" 0
-- Playing 'voucher_does_not_exist' (escape_digits=) (sample_offset
0)
<SIP/STARMG1-000003c0>AGI Tx >> 200 result=0 endpos=15200
<SIP/STARMG1-000003c0>AGI Rx << GET DATA
prepaid-voucher_enter_number
20000 9 #
-- <SIP/STARMG1-000003c0> Playing
'prepaid-voucher_enter_number.gsm'
(language 'en')
In rtp debug I notice a change in RTP sequence when voucher is being ask
for again:
Got RTP packet from x.x.x.x:22760 (type 101, seq 042355, ts 175520,
len 000004)
Got RTP RFC2833 from x.x.x.x:22760 (type 101, seq 042355, ts 175520,
len 000004, mark 0, event 00000006, end 0, duration 01120)
Got RTP packet from x.x.x.x:22760 (type 101, seq 042356, ts 175520,
len 000004)
Got RTP RFC2833 from x.x.x.x:22760 (type 101, seq 042356, ts 175520,
len 000004, mark 0, event 00000006, end 0, duration 01280)
Got RTP packet from x.x.x.x:22760 (type 101, seq 042357, ts 175520,
len 000004)
Got RTP RFC2833 from x.x.x.x:22760 (type 101, seq 042357, ts 175520,
len 000004, mark 0, event 00000006, end 0, duration 01440)
Got RTP packet from x.x.x.x:22760 (type 101, seq 042358, ts 175520,
len 000004)
Got RTP RFC2833 from x.x.x.x:22760 (type 101, seq 042358, ts 175520,
len 000004, mark 0, event 00000006, end 0, duration 01600)
Got RTP packet from x.x.x.x:22760 (type 101, seq 042359, ts 175520,
len 000004)
Got RTP RFC2833 from x.x.x.x:22760 (type 101, seq 042359, ts 175520,
len 000004, mark 0, event 00000006, end 0, duration 01760)
Got RTP packet from x.x.x.x:22760 (type 101, seq 042360, ts 175520,
len 000004)
Got RTP RFC2833 from x.x.x.x:22760 (type 101, seq 042360, ts 175520,
len 000004, mark 0, event 00000006, end 0, duration 01920)
Got RTP packet from x.x.x.x:22760 (type 101, seq 042361, ts 175520,
len 000004)
Got RTP RFC2833 from x.x.x.x:22760 (type 101, seq 042361, ts 175520,
len 000004, mark 0, event 00000006, end 1, duration 02080)
<SIP/STARMG1-000003c5>AGI Tx >> 200 result=543278456
Got RTP packet from x.x.x.x:22760 (type 101, seq 042362, ts 175520,
len 000004)
Got RTP RFC2833 from x.x.x.x:22760 (type 101, seq 042362, ts 175520,
len 000004, mark 0, event 00000006, end 1, duration 02080)
Got RTP packet from x.x.x.x:22760 (type 101, seq 042363, ts 175520,
len 000004)
Got RTP RFC2833 from x.x.x.x:22760 (type 101, seq 042363, ts 175520,
len 000004, mark 0, event 00000006, end 1, duration 02080)
Got RTP packet from x.x.x.x:22760 (type 00, seq 042364, ts 176480,
len 000160)
Got RTP packet from x.x.x.x:22760 (type 00, seq 042365, ts 176640,
len 000160)
Got RTP packet from x.x.x.x:22760 (type 00, seq 042366, ts 176800,
len 000160)
Got RTP packet from x.x.x.x:22760 (type 00, seq 042367, ts 176960,
len 000160)
<SIP/STARMG1-000003c5>AGI Rx << STREAM FILE voucher_does_not_exist
"" 0
-- Playing 'voucher_does_not_exist' (escape_digits=) (sample_offset
0)
Sent RTP packet to x.x.x.x:22760 (type 00, seq 005630, ts 151888,
len 000160)
Got RTP packet from x.x.x.x:22760 (type 00, seq 042368, ts 177120,
len 000160)
Sent RTP packet to x.x.x.x:22760 (type 00, seq 005631, ts 152048,
len 000160)
Got RTP packet from x.x.x.x:22760 (type 00, seq 042369, ts 177280,
len 000160)
Sent RTP packet to x.x.x.x:22760 (type 00, seq 005632, ts 152208,
len 000160)
Got RTP packet from x.x.x.x:22760 (type 00, seq 042370, ts 177440,
len 000160)
Sent RTP packet to x.x.x.x:22760 (type 00, seq 005633, ts 152368,
len 000160)
Got RTP packet from x.x.x.x:22760 (type 00, seq 042371, ts 177600,
len 000160)
Dave
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
<http://lists.digium.com/pipermail/asterisk-users/attachments/20120224/65222f1d/attachment.htm>
Apparently Analagous Threads
Wisdom of the Ancients
