No IVR audio. Jump in RTP sequence number

Home » Asterisk Users » No IVR audio. Jump in RTP sequence number
Asterisk Users No Comments

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@a.b.c.d SIP/2.0
Via: SIP/2.0/UDP x.x.x.x;rport;branch=z9hG4bK2vr8B7r60vS7j
Max-Forwards: 70
From: “14735201326” ;tag=0K219XHeF7K2j
To:
Call-ID: 00ddbda6-d9b1-122f-e7a7-00259025b560
CSeq: 24716447 INVITE
Contact:
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”
;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
< ------------->