One way audio inbound calls / no audio from remote

Hi, I’m having problems getting around the following problem:

One way audio inbound calls / no audio from remote

  1. I can place outbound calls without any issue, but if I receive a call, the other party can’t hear my voice…

  2. Also the inbound extension seems to ring endless after having picked up an inbound call.

Does anyone have a clue on how to get around this? I’m using freepbx with pfsense and a cisco 7961 phone

My setup:

PFSENSE firewall
(1 LAN & 1 WAN port)
external_sip_servers = all ips of my provider’s servers

  • portforwardings

wan interface


source external_sip_servers udp 5060,5061,5062
destination pbx udp 5060,5061,5062


source external_sip_servers udp 10000:20000
destination pbx udp 10000:20000

  • AON - Advanced Outbound NAT
    wan interface


source pbx udp 10000:20000
destination external_sip_servers udp 10000:20000
nat address wan address
nat port *
static port Yes


source pbx udp 5060,5061,5062
destination external_sip_servers udp 10000:20000
nat address wan address
nat port *
static port Yes


Global Settings:
  UDP Bindaddress:
  TCP SIP Bindaddress:    Disabled
  TLS SIP Bindaddress:    Disabled
  Videosupport:           No
  Textsupport:            No
  Ignore SDP sess. ver.:  No
  AutoCreate Peer:        Off
  Match Auth Username:    No
  Allow unknown access:   Yes
  Allow subscriptions:    Yes
  Allow overlap dialing:  Yes
  Allow promisc. redir:   No
  Enable call counters:   No
  SIP domain support:     No
  Realm. auth:            No
  Our auth realm          asterisk
  Use domains as realms:  No
  Call to non-local dom.: Yes
  URI user is phone no:   No
  Always auth rejects:    Yes
  Direct RTP setup:       No
  User Agent:             FPBX-13.0.74(11.22.0)
  SDP Session Name:       Asterisk PBX 11.22.0
  SDP Owner Name:         root
  Reg. context:           (not set)
  Regexten on Qualify:    No
  Trust RPID:             No
  Send RPID:              No
  Legacy userfield parse: No
  Send Diversion:         Yes
  Caller ID:              Unknown
  From: Domain:           
  Record SIP history:     Off
  Call Events:            Off
  Auth. Failure Events:   Off
  T.38 support:           No
  T.38 EC mode:           Unknown
  T.38 MaxDtgrm:          4294967295
  SIP realtime:           Disabled
  Qualify Freq :          60000 ms
  Q.850 Reason header:    No
  Store SIP_CAUSE:        No

Network QoS Settings:
  IP ToS SIP:             CS3
  IP ToS RTP audio:       EF
  IP ToS RTP video:       AF41
  IP ToS RTP text:        CS0
  802.1p CoS SIP:         4
  802.1p CoS RTP audio:   5
  802.1p CoS RTP video:   6
  802.1p CoS RTP text:    5
  Jitterbuffer enabled:   No

Network Settings:
  SIP address remapping:  Disabled
  Externhost:             <none>
  Externaddr:             (null)
  Externrefresh:          10

Global Signalling Settings:
  Codecs:                 (gsm|ulaw|alaw|g726)
  Codec Order:            ulaw:20,alaw:20,gsm:20,g726:20
  Relax DTMF:             No
  RFC2833 Compensation:   No
  Symmetric RTP:          Yes
  Compact SIP headers:    No
  RTP Keepalive:          0 (Disabled)
  RTP Timeout:            30 
  RTP Hold Timeout:       300 
  MWI NOTIFY mime type:   application/simple-message-summary
  DNS SRV lookup:         No
  Pedantic SIP support:   Yes
  Reg. min duration       60 secs
  Reg. max duration:      3600 secs
  Reg. default duration:  120 secs
  Sub. min duration       60 secs
  Sub. max duration:      3600 secs
  Outbound reg. timeout:  20 secs
  Outbound reg. attempts: 0
  Outbound reg. retry 403:0
  Notify ringing state:   Yes
    Include CID:          No
  Notify hold state:      Yes
  SIP Transfer mode:      open
  Max Call Bitrate:       384 kbps
  Auto-Framing:           No
  Outb. proxy:            <not set> 
  Session Timers:         Accept
  Session Refresher:      uas
  Session Expires:        1800 secs
  Session Min-SE:         90 secs
  Timer T1:               500
  Timer T1 minimum:       100
  Timer B:                32000
  No premature media:     Yes
  Max forwards:           70

Default Settings:
  Allowed transports:     UDP
  Outbound transport:      UDP
  Context:                from-sip-external
  Record on feature:      automon
  Record off feature:     automon
  Force rport:            Yes
  DTMF:                   rfc2833
  Qualify:                0
  Keepalive:              0
  Use ClientCode:         No
  Progress inband:        Never
  Language:               de
  Tone zone:              <Not set>
  MOH Interpret:          default
  MOH Suggest:            
  Voice Mail Extension:   *97


Usually this is a nat issue. Do you register to a voip provider to receive calls or the calls that you are getting are from remote extensions?

[quote=“astbox, post:2, topic:35484”]
Do you register to a voip provider to receive calls
[/quote] yes I’m registering to a voip provider.

Addendum: The mentioned behavior (one way audio) seems to be dependent to the provider of the calling party… On countless tests it worked flawless, but then…

Maybe your provider has a routing issue if you have problem with calls from specific providers. Make a sip trace to see what is happeing when you receive a call.


Looks like your PBX is behind nAT but not setup to be behind NAT. Make sure your PBX is setup properly to show it’s behind NAT. You’re not sending proper information in your SIP messages which is mostly either not making it anywhere or causing the destination to send replies back to the wrong IP or get lost in your NAT tables.