Asterisk PJSIP: Provider Outbound Call (Latest Versions)

Outbound calling through a SIP provider is one of the most business-critical parts of any Asterisk deployment.

Even when inbound calls work perfectly, outbound failures are common due to:

• Dialplan routing mistakes
• Caller ID rejection by provider
• Codec negotiation problems
• NAT or firewall blocking RTP media
• Provider policy limits or number formatting errors

This guide explains the complete outbound call path using modern PJSIP trunks and shows how to debug every real-world failure scenario.

Outbound Call Flow (Real Telecom Mental Model)

1. An extension, IVR, or dialer triggers a Dial() command
2. Asterisk selects the correct PJSIP endpoint (trunk)
3. Asterisk sends SIP INVITE to the provider
4. Provider validates:
   • Source IP or registration
   • Caller ID permissions
   • Dialed number format
5. Provider connects the call to PSTN/mobile network
6. RTP audio flows between Asterisk and provider gateway

Failure can occur at any layer of this chain.

Basic Outbound Dialplan Pattern

A clean outbound design always:

• Logs the call attempt
• Normalizes number format
• Sets correct caller ID
• Handles provider failure gracefully

[outbound]
exten => _X.,1,NoOp(Outbound attempt: ${EXTEN} from ${CALLERID(all)})

; Normalize number if needed (example: ensure country code)
 same => n,Set(DIALNUM=${EXTEN})

; Set approved caller ID (must be allowed by provider)
 same => n,Set(CALLERID(num)=1800123456)

; Place outbound call
 same => n,Dial(PJSIP/provider-endpoint/${DIALNUM},60)

; If call fails, log cause
 same => n,NoOp(Dial status: ${DIALSTATUS}  Hangup cause: ${HANGUPCAUSE})
 same => n,Hangup()

PJSIP Endpoint Requirements for Outbound Calls

Your provider endpoint must allow outbound media and signaling stability:

[provider-endpoint]
type=endpoint
transport=transport-udp
disallow=all
allow=ulaw,alaw

aors=provider-aor
outbound_auth=provider-auth   ; only for registration-based trunks

; NAT/media stability
direct_media=no
rtp_symmetric=yes
force_rport=yes
rewrite_contact=yes

Incorrect codec or NAT settings are a major cause of silent audio.

Number Formatting Problems (Most Common Real Failure)

Providers expect specific dial formats such as:

• +919876543210 (E.164)
• 00919876543210 (international prefix)
• 9876543210 (national format)

If the number format is wrong, providers may return:

• 404 Not Found
• 484 Address Incomplete
• 403 Forbidden

Always confirm the required dial format with your provider.

Caller ID Rejection (Very Common in Production)

Providers often restrict which caller IDs you can present.

Symptoms:

• Outbound call fails instantly
• SIP response shows 403 Forbidden

Fix:

• Use a provider-approved DID
• Set it in dialplan or endpoint from_user

Codec Negotiation Failures

If provider supports only PCMU/PCMA but endpoint allows Opus/GSM, the call may fail with:

• 488 Not Acceptable Here
• Immediate call drop after answer

Restrict codecs explicitly:

disallow=all
allow=ulaw,alaw

NAT and RTP Audio Issues in Outbound Calls

Real-world symptom:

• Call connects successfully
• But one-way or no audio occurs

Root cause is usually:

• Firewall blocking RTP ports
• Incorrect external_media_address

Required firewall rules:

sudo ufw allow 5060/udp
sudo ufw allow 10000:20000/udp

Debugging Outbound Failures Step-by-Step

1. Enable SIP logging

pjsip set logger on

Look for provider response codes:

• 403 → caller ID or permission issue
• 404/484 → wrong number format
• 488 → codec mismatch
• No response → firewall/NAT/network

2. Confirm RTP flow

rtp set debug on

3. Capture packets if unclear

sudo tcpdump -i any -s 0 -w outbound.pcap udp port 5060 or udp portrange 10000-20000

Analyze in Wireshark to confirm SIP + RTP behavior.

Handling Provider Failures Gracefully

Production dialplans should always handle failure paths:

same => n,GotoIf($["${DIALSTATUS}"="BUSY"]?busy)
same => n,GotoIf($["${DIALSTATUS}"="NOANSWER"]?noanswer)
same => n,GotoIf($["${DIALSTATUS}"="CONGESTION"]?congestion)

same => n(busy),Playback(line-busy)
same => n,Hangup()

same => n(noanswer),Playback(vm-nobodyavail)
same => n,Hangup()

same => n(congestion),Playback(all-circuits-busy-now)
same => n,Hangup()

Production Checklist for Reliable Outbound Calling

• Correct number normalization format
• Provider-approved caller ID configured
• Allowed codecs match provider
• Firewall allows SIP + RTP
• NAT external IP configured properly
• SIP logging + tcpdump workflow documented

Key Takeaway

Outbound calling reliability depends on understanding the full telecom chain:

• Dialplan logic
• Provider policies
• Codec compatibility
• Network/NAT behavior

When you debug outbound calls layer-by-layer instead of guessing, even complex provider failures become predictable and solvable.