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