Outbound SIP Call – No Caller Audio Reaching Asterisk (One-Way Audio Issue)

Asterisk Asterisk SIP
1

Hello Asterisk Community,
We are experiencing a consistent one-way audio issue when placing outbound calls from Asterisk to Twilio SIP Trunk and then to the PSTN.

Problem Summary

  • Outbound calls are established successfully.

  • Audio from Asterisk → Twilio → Callee works correctly (TTS prompts are heard).

  • Audio from Caller → Twilio → Asterisk is not received.

  • Asterisk receives no inbound RTP packets.

  • ARI application generates recordings with only a 44-byte WAV header, indicating no audio data.

Example Recording Log

  • Processing recording file: 44 bytes

  • OUTBOUND CALL DIAGNOSIS: Empty recording (44-byte WAV header only)

  • Confirms no inbound audio is reaching Asterisk.

Audio Path Summary

  • Asterisk → Twilio → Callee: Working

  • Caller → Twilio → Asterisk: Not working

  • Indicates possible NAT, firewall, or RTP traversal issue.

Environment Details

  • Asterisk version: 22.6.0

  • OS: Ubuntu 22.04.5 LTS, Kernel 6.8.0-87-generic

  • Deployment: KVM Virtual Machine

  • Firewall status:

    • UFW: inactive

    • iptables: UDP 5060 and UDP 10000–20000 allowed

  • Public IP: 125.22.172.57

  • Device behind NAT: Yes

  • Twilio SIP trunking: UDP, SIP signaling range 54.172.60.0/23

Relevant PJSIP Configuration

Simplified Endpoint Section

  • direct_media=no

  • rtp_symmetric=yes

  • force_rport=yes

  • rewrite_contact=yes

Full PJSIP Configuration

Endpoint

  • type=endpoint

  • transport=0.0.0.0-udp

  • context=from-trunk

  • disallow=all

  • allow=ulaw,alaw,gsm,g726,g722,h264,mpeg4

  • aors=Twilio-Out

  • outbound_auth=Twilio-Out

  • from_domain=microgrid.pstn.twilio.com

  • from_user=17372323899

  • trust_id_inbound=yes

  • send_pai=yes

  • direct_media=no

  • rewrite_contact=yes

  • rtp_symmetric=yes

  • dtmf_mode=rfc4733

  • media_address=49.204.231.27

  • language=en

AOR

  • contact=sip:freepbxsip@microgrid.pstn.twilio.com:5060

  • qualify_frequency=60

  • default_expiration=3600

  • maximum_expiration=7200

  • minimum_expiration=60

Auth

  • auth_type=userpass

  • username=freepbxsip

  • password=Microgrid@123

Identify

  • endpoint=Twilio-Out

  • match=54.172.60.0/23

Request for Assistance

We are trying to determine why Asterisk is not receiving inbound RTP from Twilio despite:

  • Outbound RTP working correctly.

  • SIP/SDP negotiation appearing normal.

  • Twilio IP ranges configured properly.

  • RTP/SIP ports opened in firewall.

  • No inbound RTP visible using rtp set debug on.

What We Need Help With

  • Additional logs required?

  • Specific SIP traces?

  • Packet captures (tcpdump)?

  • Any configuration we should adjust or verify?

  • Any known Twilio/Asterisk NAT or media-handling considerations?

2

The likely error (missing external media and local networks entries) is in the type=transport section, which you haven’t provided.

The last three”simplified endpoint” options are not relevant, if I correctly understand your NAT environment; they are for when the endpoint, itself, is behind NAT, and doesn’t compensate for that. As you only have one leg, the direct_media setting is not relevant.

3

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