Incoming sip rings, but no voice

Asterisk Asterisk Support
1

I dial my fwd num on ekiga, asterisk answers and rings the extension. I can answer, but no voice, noise, hum or anything.

CLI output:

[Oct 8 15:41:31] NOTICE[29102]: chan_iax2.c:6787 socket_process: Rejected connect attempt from 192.246.69.186, who was trying to reach ‘@’
– Executing [s@FWDincoming:1] Answer(“SIP/fwd.pulver.com-0066e710”, “”) in new stack
– Executing [s@FWDincoming:2] Dial(“SIP/fwd.pulver.com-0066e710”, “Zap/1”) in new stack
– Called 1>
– Zap/1-1 is ringing
– Zap/1-1 is ringing
– Zap/1-1 is ringing
– Zap/1-1 is ringing
– Zap/1-1 is ringing
– Zap/1-1 answered SIP/fwd.pulver.com-0066e710
– Hungup ‘Zap/1-1’

What’s the rejection notice?? I don’t what the 192.246.69.186 ip address is.

sip.conf
[general]
context=default
srvlookup=yes
realm = asterisk
;bindport = 5061
language=en
disallow=all
allow = gsm ; what we deem is necessary
allow = ilbc
allow = speex
;allow = g729 ; g729 only works for pass-thru, if you haven’t bought a license
allow = g726
allow = ulaw
tos_sip = cs3
tos_audio = ef
tos_video = af41
register => <FWD#>:<FWD_PASSWORD>@fwd.pulver.com
context=FWDincoming
nat=yes
insecure=port,invite
externhost=myphone.dtdns.net
localnet=192.168.0.0/255.255.0.0

extensions.conf
[FWDincoming]
exten => s,1,Answer()
exten => s,2,Dial(Zap/1)

sean

2

Took out the extraneous context=default at top of sip.conf and commented out allow=speex. Still same result : ring but no voice

set sip debug:

…
– Executing [s@FWDincoming:1] Answer(“SIP/fwd.pulver.com-0065d050”, “”) in new stack
Audio is at 69.183.180.190 port 12948
Adding codec 0x2 (gsm) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 69.90.155.70:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 69.90.155.70;branch=z9hG4bK831d.d1c0e147.1;received=69.90.155.70
Via: SIP/2.0/UDP 69.183.180.190:5099;branch=z9hG4bKac32985d-7655-db11-8c6c-0015f2778613;rport=5099
Record-Route: sip:69.90.155.70;ftag=0cd2ee5c-7655-db11-8c6c-0015f2778613;lr=on
…
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: sip:s@69.183.180.190
Content-Type: application/sdp
Content-Length: 279

v=0
o=root 29082 29082 IN IP4 69.183.180.190
s=session
c=IN IP4 69.183.180.190
t=0 0
m=audio 12948 RTP/AVP 3 0 101
a=rtpmap:3 GSM/8000
a=ptime:20
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=sendrecv
mediaserver*CLI>

-- Executing [s@FWDincoming:2] Dial("SIP/fwd.pulver.com-0065d050", "Zap/1") in new stack
-- Called 1> 
-- Zap/1-1 is ringing

mediaserver*CLI>
<-- SIP read from 69.90.155.70:5060:
ACK sip:s@69.183.180.190 SIP/2.0
Record-Route: sip:69.90.155.70;ftag=0cd2ee5c-7655-db11-8c6c-0015f2778613;lr=on
CSeq: 2 ACK
Via: SIP/2.0/UDP 69.90.155.70;branch=0
Via: SIP/2.0/UDP 69.183.180.190:5099;branch=z9hG4bKf48a475e-7655-db11-8c6c-0015f2778613;rport=5099
…
Contact: sip:681526@69.183.180.190:5099;transport=udp
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,NOTIFY,REFER,MESSAGE,INFO,PING
Content-Length: 0
Max-Forwards: 16

— (12 headers 0 lines)—
– Zap/1-1 is ringing
– Zap/1-1 answered SIP/fwd.pulver.com-0065d050

??

sean

3

  1. ensure externip= and localnet= in sip.conf are correct.

  2. ensure that you have your rtp ports (any range, define it in rtp.conf) non-firewalled and forwarded to the * box (this si probably your problem)

  3. try rtp debug and see if you get anything…