Asterisk doesnt recognize DTMF and send error

Asterisk Asterisk Support
1

Hi. I’m using Asterisk v1.4.18 on CentOS 4.5
I have problem when I make the call using X-Lite. Here it is

[Mar 24 00:09:28] WARNING[6828]: chan_sip.c:1947 retrans_pkt: Maximum retries exceeded on transmission ***********. for seqno 2 (Critical Response)
[Mar 24 00:09:28] WARNING[6828]: chan_sip.c:1971 retrans_pkt: Hanging up call ***********. - no reply to our critical packet.

When I use sip debug. I saw that the transmission repeats 6 time. Asterisk doesn’t detect the DTMF that I pressed.

.........(first 5 times)

Retransmitting #6 (no NAT) to **************:
SIP/2.0 200 OK
Via: SIP/2.0/UDP **************:28180:50058;branch=z9hG4bK-d8754z-f711081e7455530e-1---d8754z-;received=************;rport=28180
From: "102"<sip:102@***********>;tag=f448dd55
To: "19001773"<sip:19001773@**************>;tag=as26d5501d
Call-ID: ***************.
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:19001773@**************>
Content-Type: application/sdp
Content-Length: 262

v=0
o=root 6732 6732 IN IP4 **************
s=session
c=IN IP4 **************
t=0 0
m=audio 12230 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

Here is my sip.conf

[general]
context=default			; Default context for incoming calls
;allowguest=no			; Allow or reject guest calls (default is yes)
allowoverlap=no			; Disable overlap dialing support. (Default is yes)
;allowtransfer=no		; Disable all transfers (unless enabled in peers or users)
				; Default is enabled
;realm=mydomain.tld		; Realm for digest authentication
				; defaults to "asterisk". If you set a system name in
				; asterisk.conf, it defaults to that system name
				; Realms MUST be globally unique according to RFC 3261
				; Set this to your host name or domain name
bindport=5060			; UDP Port to bind to (SIP standard port is 5060)
				; bindport is the local UDP port that Asterisk will listen on
bindaddr=0.0.0.0		; IP address to bind to (0.0.0.0 binds to all)
srvlookup=yes			; Enable DNS SRV lookups on outbound calls
				; Note: Asterisk only uses the first host 
				; in SRV records
				; Disabling DNS SRV lookups disables the 
				; ability to place SIP calls based on domain 
				; names to some other SIP users on the Internet

[101]
type=friend
username=101
secret=123456
context=from-sip
regexten=1234			; When they register, create extension 1234
dtmfmode=rfc2833
host=dynamic			; This device needs to register
nat=no			; X-Lite is behind a NAT router
canreinvite=no			; Typically set to NO if behind NAT
disallow=all
allow=gsm			; GSM consumes far less bandwidth than ulaw
allow=ulaw
allow=alaw

[102]
type=friend
username=102
secret=123456
context=from-sip
regexten=1234			; When they register, create extension 1234
dtmfmode=rfc2833
host=dynamic			; This device needs to register
nat=no			; X-Lite is behind a NAT router
canreinvite=no			; Typically set to NO if behind NAT
disallow=all
allow=gsm			; GSM consumes far less bandwidth than ulaw
allow=ulaw
allow=alaw

I’ve google but I cannot find any answer. Plz help me.

2

Asterisk cannot recognize DTMF until after the call has been answered. Asterisk is trying to answer the call but the phone is failing to acknowledge, which probably means you have a NAT problem.

Did you really mean DTMF, or did you mean the dailled digits. The latter should be the 19001773 in the To: headers.

3

Yes. I open 10000-20000 port in my modem. It is used for rpt protocol.
thank you for help.

4

Failing to answer indicate a SIP issue, not an RTP one. Opening ports is only sufficient if you are using public addresses internally.