Providers

• Overview
• Shared Provider Fields
• SIP Provider
• PJSIP Provider
• IAX2 Provider
• DAHDI Provider
• LOCAL Provider
• SMS Provider

Overview

Providers are global carrier trunks or external services used by routing profiles, inbound DIDs, SMS delivery, and cost handling. Use Admin > Providers to create SIP, PJSIP, IAX2, DAHDI, LOCAL, or SMS providers and to review live channel counters.

Technology Selection

Technology	Use for	Typical fictional example
SIP	Legacy chan_sip carrier trunks or IP-authenticated SIP gateways.	Docs Telecom SIP Trunk at 203.0.113.20
PJSIP	Modern res_pjsip carrier trunks with endpoint, AOR, auth, and identify rows.	Docs Telecom PJSIP Trunk with TLS/SRTP
IAX2	IAX2 gateways or legacy interconnects.	Docs IAX2 Gateway on port 4569
DAHDI	Analog, PRI, or hardware channel-group routing controlled outside SIP/PJSIP realtime.	Docs Analog Gateway
LOCAL	Local PBX routing targets that should be selected like a provider.	Docs Local Route
SMS	SMS delivery via HTTP, JSON, SOAP, SIP MESSAGE, SMSPortal, or AGI script.	Docs SMS JSON Gateway

Configuration Workflow

• Create or open a provider from Admin > Providers.
• Complete the shared identity, outbound, and inbound fields that apply to the selected technology.
• For SIP, PJSIP, or IAX2, enable Use Realtime Account when MiRTA PBX should maintain the Asterisk realtime rows.
• Save the provider only after replacing the fictional documentation values with carrier-supplied production values.
• Assign providers to routing profiles and test inbound DID matching, outbound dialing, caller ID presentation, and SMS delivery.

Operational Notes

• Provider changes are global and can affect multiple tenants, routing profiles, DIDs, and SMS routes.
• Use documentation-reserved IP addresses and example domains only for training material; never use them in production trunks.
• Provider-level CallerID and CallerID Name override extension-derived caller ID for every call through the provider.
• Max outbound channels prevents a trunk from being overused. When the limit is reached, routing can continue to another provider.

Shared Provider Fields

These fields appear on the provider page before technology-specific realtime options. Some fields are hidden when they do not apply to the selected provider technology.

Information

Field	Applies to	Purpose	Fictional example
Name	All technologies	Friendly provider name shown in provider lists, route choices, reports, and provider-cost screens.	Docs Telecom SIP Trunk
Peer Name	All technologies	Stable technical identifier used by routing and, for realtime accounts, by Asterisk peer or endpoint rows. Keep it unique and avoid spaces.	docs_telecom_sip
Technology	All technologies	Selects the provider driver or service type. It controls which realtime and SMS fields are visible.	SIP, PJSIP, IAX2, DAHDI, LOCAL, or SMS
Status	All technologies	Controls whether the provider is usable. Enabled allows inbound and outbound use, Disabled removes it from routing, Only Inbound keeps inbound handling only.	Enabled
Provider Costs	Existing providers	Opens the provider-cost editor for rating or cost data associated with the provider.	Open costs for Docs Telecom SIP Trunk

Outbound Calls

Field	Applies to	Purpose	Fictional example
Additional Header 1-6	SIP, PJSIP, IAX2, SMS	Adds SIP headers or SMS HTTP headers. Use one header per field in Header-Name: value format.	X-Account-Code: docs-demo
CallerID Presentation	SIP, PJSIP, IAX2, DAHDI, LOCAL	Controls privacy and screening information sent when routing calls through the provider.	Auto Privacy, CallerID not screened
Digits to Add	All outbound providers	Prepends digits to the called number before the call or SMS is sent to the provider.	+1
Number of digits to Remove	All outbound providers	Removes leading digits from the called number before adding any outbound prefix.	1
CallerID Modifications	Voice providers	Optional outbound caller ID modification rule applied before the call leaves through this provider.	No Modifications or Docs US E164 CallerID
CallerID	Voice providers	Forces a caller ID number for every call using the provider. Leave blank to use extension-derived caller ID.	+15551230000
CallerID Name	Voice providers	Forces a caller ID name for every call using the provider. Leave blank to use extension-derived caller ID name.	Docs Main
CallerID allowed regex	Voice providers	Validates caller ID numbers accepted by the carrier. If the current caller ID does not match, MiRTA PBX falls back to DID, extension, or tenant defaults.	^\+1555[0-9]{7}$
Max outbound channels	All outbound providers	Limits concurrent outbound channels. When the limit is reached, routing skips to another provider. Use Unlimited for no limit.	12
Penalty	All outbound providers	Provider selection weight used by routing and round-robin choices. Lower penalty is preferred.	10
Ignore SIP Cause codes	Voice providers	Continues to another provider even when this provider returns a SIP failure cause.	checked for fallback routing
Ignore BUSY	Voice providers	Continues to another provider when this provider returns BUSY and the route should keep searching.	checked for multi-carrier routes

Inbound Calls

Field	Applies to	Purpose	Fictional example
Inbound Digits to Add	Inbound voice providers	Prepends digits to the inbound DID or called number before destination matching.	+1
Inbound Number of digits to Remove	Inbound voice providers	Removes leading digits from the inbound DID or called number before adding any inbound prefix.	0
DID Modifications	Inbound voice providers	Optional modification rule applied to inbound called numbers before DID matching.	Docs US DID Normalize
Same provider IN and OUT	Inbound and forwarded voice calls	Controls whether calls entering from the provider may also be sent out through the same provider.	Yes
Allow custom extensions	Inbound voice providers	Allows inbound calls from the provider to match custom extensions. Use only for trusted providers.	unchecked
Normalize number	Inbound voice providers	Removes URI or signaling decoration before matching the number to PBX destinations.	checked
Get number from	Inbound SIP/PJSIP-style signaling	Chooses where the inbound called number is read from: standard Request-URI, SIP To header, or a fixed value.	SIP To: Header
Set to	When Get number from is Set to	Fixed DID or destination number used instead of a number from signaling.	+15551239999
Variable name 1-3	Inbound voice providers	Custom inbound channel variable name stored with the USR- prefix for routing or dialplan logic.	ProviderName
Variable value 1-3	Inbound voice providers	Value assigned to the matching custom inbound channel variable.	Docs Telecom SIP Trunk

Save, Delete, and Back

Field	Applies to	Purpose	Fictional example
Save	All providers	Stores the provider and creates, updates, or removes realtime account rows as required by the selected technology and Use Realtime Account.	Save after replacing all fictional values.
Delete	Existing providers	Deletes the provider and removes related realtime rows for SIP, PJSIP, or IAX2 providers.	Delete an unused Docs Test Provider after checking routing references.
Back	All providers	Returns to the Providers list without submitting the form.	Use after reviewing settings.
Floating Save	Long forms	Lets administrators save from lower sections without scrolling back to the bottom button row.	Visible while editing realtime account fields.

SIP Provider

Use SIP technology for chan_sip provider trunks. SIP providers can create and maintain a global sipfriends realtime row when Use Realtime Account is enabled.

Common Realtime Fields

Field	Applies to	Purpose	Fictional example
Use Realtime Account	SIP, PJSIP, IAX2	Creates and maintains the Asterisk realtime peer, endpoint, AOR, auth, or IAX row for the provider.	checked
Transport	SIP, PJSIP	SIP signaling transport. For PJSIP this maps to transport objects such as transport-udp, transport-tcp, or transport-tls.	UDP or TLS
RTP Encryption (SRTP)	SIP, PJSIP	Enables SRTP media encryption when supported by the provider. PJSIP stores this as media_encryption=sdes.	No for standard UDP SIP, Yes for TLS/SRTP trunks
Host	SIP, PJSIP, IAX2	Remote provider address. Use an IP address when inbound realtime matching must identify the carrier by source IP.	203.0.113.20
Port	SIP, PJSIP, IAX2	Remote signaling port. Defaults are commonly 5060 for SIP/PJSIP and 4569 for IAX2.	5060
Outbound Proxy	SIP, PJSIP	Proxy URI or host used for outbound signaling when required by the carrier.	sip-proxy.example.invalid
Username	SIP, PJSIP, IAX2	Authentication username sent to the provider. PJSIP stores this in ps_auths.username, chan_sip as defaultuser.	docs_sip_user
Password	SIP, PJSIP, IAX2	Authentication secret for the provider. Use a unique carrier-supplied or generated secret.	FictionalSecret-2026
Qualify	SIP, PJSIP	Enables SIP OPTIONS reachability checks and controls timeout behavior.	Yes
Qualify Frequency	SIP, PJSIP	Interval in seconds between provider reachability checks.	60 secs
Codecs	SIP, PJSIP, IAX2	Allowed audio codecs. MiRTA PBX stores disallow=all and the selected allow list.	alaw, ulaw, g722
Session Expires	SIP, PJSIP	Maximum SIP session refresh interval in seconds.	1800
DTMF Mode	SIP, PJSIP	DTMF signaling method. PJSIP maps RFC 2833 to rfc4733.	RFC 2833/RFC 4733
RTP Keepalive	SIP, PJSIP	Sends RTP keepalive packets to help keep NAT bindings and media paths active.	5 seconds

SIP-Only Fields

Field	Applies to	Purpose	Fictional example
From User	SIP only	chan_sip fromuser value used in the SIP From header.	+15551230000
From Domain	SIP only	chan_sip fromdomain value used when the provider requires a specific domain in the From header.	sip.example.invalid
Send RPID	SIP only	Controls whether caller identity is sent using Remote-Party-ID, P-Asserted-Identity, or both.	PAI
Trust RPID	SIP only	Controls whether inbound Remote-Party-ID or P-Asserted-Identity data is trusted for caller identity.	PAI
Trust Outbound for CallerID Presentation	SIP only	Controls how prohibited caller presentation is handled in RPID/PAI headers.	Legacy
Session Timers	SIP only	chan_sip session timer policy: accept remote requests, originate timers, or refuse timers.	Accept
Progress inband	SIP only	Controls whether Asterisk generates in-band progress audio before answer.	Yes
NAT	SIP only	chan_sip NAT behavior. force_rport,comedia is common for symmetric RTP and response routing.	force_rport,comedia
Call response timer (ms) - T1	SIP only	Milliseconds to wait for response to SIP messages.	500
Call setup timer (ms) - B	SIP only	Maximum milliseconds to wait for call setup before autocongestion when no provisional response arrives.	5000
Can Reinvite	SIP only	Allows Asterisk to attempt direct media re-invites. Use No when recording, NAT, or media control should keep Asterisk in the RTP path.	No
Allow Transfer	SIP only	chan_sip allowtransfer. When enabled, the peer may request SIP REFER transfers.	No
Insecure	SIP only	chan_sip authentication matching option. Port, Invite is common for IP-authenticated trunks.	Port, Invite

Fictional SIP Example

Name: Docs Telecom SIP Trunk
Peer Name: docs_telecom_sip
Technology: SIP
Host: 203.0.113.20
Port: 5060
Username: docs_sip_user
From User: +15551230000
From Domain: sip.example.invalid
NAT: force_rport,comedia
Insecure: Port, Invite

PJSIP Provider

Use PJSIP technology for res_pjsip provider trunks. MiRTA PBX maintains related ps_endpoints, ps_aors, ps_auths, and ps_endpoint_id_ips rows when realtime is enabled.

Common Realtime Fields

Field	Applies to	Purpose	Fictional example
Use Realtime Account	SIP, PJSIP, IAX2	Creates and maintains the Asterisk realtime peer, endpoint, AOR, auth, or IAX row for the provider.	checked
Transport	SIP, PJSIP	SIP signaling transport. For PJSIP this maps to transport objects such as transport-udp, transport-tcp, or transport-tls.	UDP or TLS
RTP Encryption (SRTP)	SIP, PJSIP	Enables SRTP media encryption when supported by the provider. PJSIP stores this as media_encryption=sdes.	No for standard UDP SIP, Yes for TLS/SRTP trunks
Host	SIP, PJSIP, IAX2	Remote provider address. Use an IP address when inbound realtime matching must identify the carrier by source IP.	203.0.113.20
Port	SIP, PJSIP, IAX2	Remote signaling port. Defaults are commonly 5060 for SIP/PJSIP and 4569 for IAX2.	5060
Outbound Proxy	SIP, PJSIP	Proxy URI or host used for outbound signaling when required by the carrier.	sip-proxy.example.invalid
Username	SIP, PJSIP, IAX2	Authentication username sent to the provider. PJSIP stores this in ps_auths.username, chan_sip as defaultuser.	docs_sip_user
Password	SIP, PJSIP, IAX2	Authentication secret for the provider. Use a unique carrier-supplied or generated secret.	FictionalSecret-2026
Qualify	SIP, PJSIP	Enables SIP OPTIONS reachability checks and controls timeout behavior.	Yes
Qualify Frequency	SIP, PJSIP	Interval in seconds between provider reachability checks.	60 secs
Codecs	SIP, PJSIP, IAX2	Allowed audio codecs. MiRTA PBX stores disallow=all and the selected allow list.	alaw, ulaw, g722
Session Expires	SIP, PJSIP	Maximum SIP session refresh interval in seconds.	1800
DTMF Mode	SIP, PJSIP	DTMF signaling method. PJSIP maps RFC 2833 to rfc4733.	RFC 2833/RFC 4733
RTP Keepalive	SIP, PJSIP	Sends RTP keepalive packets to help keep NAT bindings and media paths active.	5 seconds

PJSIP-Specific Fields

Field	Applies to	Purpose	Fictional example
Transport	PJSIP only	Maps to the selected PJSIP transport object. Use TLS when the carrier requires encrypted signaling.	TLS
Host and Port	PJSIP only	Used to build the AOR contact and, when Host is an IP address, the endpoint identify match.	198.51.100.30:5061
Username and Password	PJSIP only	Stored in ps_auths. If username is empty, MiRTA PBX does not attach username/password auth.	docs_pjsip_user
Outbound Proxy	PJSIP only	PJSIP outbound proxy string for the endpoint.	sip:edge.example.invalid;transport=tls
Send RPID	PJSIP only	Maps to PJSIP send_rpid and send_pai.	PAI
Direct Media	PJSIP only	Allows PJSIP to attempt direct RTP. Disable when NAT, recording, transcoding, or media control requires Asterisk in the RTP path.	No
Allow Transfer	PJSIP only	PJSIP allow_transfer. When enabled, the endpoint may request SIP REFER transfers.	No
RTP Encryption (SRTP)	PJSIP only	Stores media_encryption=sdes when enabled.	Yes
Qualify Frequency and Timeout	PJSIP only	Stored on the AOR as qualify_frequency and qualify_timeout.	60 secs

Fictional PJSIP Example

Name: Docs Telecom PJSIP Trunk
Peer Name: docs_telecom_pjsip
Technology: PJSIP
Transport: TLS
RTP Encryption: Yes
Host: 198.51.100.30
Port: 5061
Outbound Proxy: sip:edge.example.invalid;transport=tls
Send RPID: PAI
Direct Media: No
Allow Transfer: No

IAX2 Provider

Use IAX2 technology for IAX trunks and legacy interconnects. IAX2 uses fewer signaling fields than SIP/PJSIP and stores its realtime account in iaxfriends.

IAX2 Fields

Field	Applies to	Purpose	Fictional example
Host	IAX2 only	Remote IAX2 peer address.	203.0.113.45
Port	IAX2 only	Remote IAX2 signaling/media port.	4569
Username	IAX2 only	Stored as the IAX2 username used for authentication.	docs_iax_user
Password	IAX2 only	IAX2 secret used for authentication.	FictionalSecret-2026
Codecs	IAX2 only	Allowed codecs stored in the IAX realtime row.	alaw, ulaw
Use Realtime Account	IAX2 only	Creates or updates the row in iaxfriends.	checked

Shared Routing Fields

IAX2 providers still use the shared outbound and inbound fields for digit manipulation, caller ID presentation, caller ID fallback, channel limits, penalties, and custom inbound variables.

Fictional IAX2 Example

Name: Docs IAX2 Gateway
Peer Name: docs_iax2_gateway
Technology: IAX2
Host: 203.0.113.45
Port: 4569
Username: docs_iax_user
Codecs: alaw, ulaw

DAHDI Provider

Use DAHDI technology for analog, PRI, or hardware channel-group resources that are managed by the telephony node outside SIP/PJSIP realtime. DAHDI providers do not show the realtime account block.

DAHDI Field Details

Field	Applies to	Purpose	Fictional example
Technology	DAHDI only	Use when routing to analog, PRI, or channel-group resources managed by DAHDI outside SIP/PJSIP realtime.	DAHDI
Peer Name	DAHDI only	Identifier operators recognize as the DAHDI group or channel target.	dahdi_docs_group1
CallerID Presentation	DAHDI only	Controls presentation flags applied before dialing through the DAHDI provider.	Auto Privacy and Screen
Digits to Add / Remove	DAHDI only	Normalizes outbound numbers for PSTN gateway expectations.	Remove 0, add blank
Get number from / Set to	Inbound DAHDI use	Sets or normalizes the inbound called number when signaling does not carry the DID in a SIP-style header.	Set to +15551234444
Max outbound channels	DAHDI only	Protects limited analog or PRI capacity.	4

Shared Fields Still Used

DAHDI providers still use provider name, peer name, status, outbound digit manipulation, caller ID presentation, caller ID overrides, caller ID allowed regex, max outbound channels, penalty, inbound digit manipulation, DID modifications, custom variables, and save/delete controls.

Fictional DAHDI Example

Name: Docs Analog Gateway
Peer Name: dahdi_docs_group1
Technology: DAHDI
Status: Enabled
Max outbound channels: 4
Get number from: Set to +15551234444

LOCAL Provider

Use LOCAL technology when a PBX-local route should be selected by routing profiles like a provider. LOCAL providers do not create SIP, PJSIP, or IAX realtime rows.

LOCAL Field Details

Field	Applies to	Purpose	Fictional example
Technology	LOCAL only	Uses a local route target instead of an external SIP/IAX/DAHDI carrier account.	LOCAL
Peer Name	LOCAL only	Stable local-route identifier for administrators and routing profiles.	docs_local_route
Digits to Add / Remove	LOCAL only	Normalizes the number before local routing.	No add, remove 0
CallerID and CallerID Name	LOCAL only	Optionally forces caller ID for calls sent through the local route.	+15551230000 and Docs Local
Max outbound channels	LOCAL only	Limits local-route concurrency, or Unlimited when no limit is needed.	Unlimited
Inbound options	LOCAL when used for inbound-like routing	Can still normalize destination numbers and set custom variables before matching.	Normalize number checked

Shared Fields Still Used

LOCAL providers still use common identity, status, outbound digit manipulation, caller ID controls, max outbound channels, penalty, inbound normalization, DID modification, and custom variable fields. Additional SIP/SMS headers and realtime account settings are hidden.

Fictional LOCAL Example

Name: Docs Local Route
Peer Name: docs_local_route
Technology: LOCAL
Status: Enabled
Max outbound channels: Unlimited
Penalty: 50

SMS Provider

Use SMS technology for outbound text-message delivery. SMS providers reuse provider identity, status, outbound digit manipulation, additional headers, channel limits, and penalty fields, then expose protocol-specific delivery fields.

SMS Field Details

Field	Applies to	Purpose	Fictional example
SMS Protocol	SMS only	Selects delivery method: Web URL, JSON Web URL, SOAP Web URL, SIP MESSAGE, SMSPORTAL, or AGISCRIPT.	JSONWEBURL
SIP Provider	SMS SIPMESSAGE	SIP provider used to send outbound SMS with SIP MESSAGE.	Docs Telecom SIP Trunk
SMS URL/AGI Script Name	WEBURL, JSONWEBURL, SOAPWEBURL, AGISCRIPT	HTTP endpoint or AGI script name used for delivery.	https://sms.example.invalid/v1/messages
SOAP Action	SOAPWEBURL	SOAPAction header value sent with SOAP SMS requests.	SendMessage
HTTP Content Type	JSONWEBURL, SOAPWEBURL	Content-Type header. Leave blank to use the protocol default when acceptable.	application/json or text/xml
User / Password	WEBURL, JSONWEBURL, SOAPWEBURL	HTTP authentication credentials when required by the SMS provider.	docs-json-user and a fictional secret
Client ID / API Secret	SMSPORTAL	SMSPortal credentials. MiRTA PBX uses them to request a bearer token before sending messages.	fictional-client-id
Post data	WEBURL	POST body in param=value&param2=value format. Values are not automatically URL encoded.	to=${SMSDESTNUM}&text=${URIENCODE(${SMSTEXT})}
JSON data	JSONWEBURL	One field.path:value[:type] line per JSON field. Dot notation creates nested objects; numeric parts can create JSON arrays.	messages.0.content:${SMSTEXT}
SOAP data	SOAPWEBURL	SOAP or XML body sent to the SMS provider.	<Text>${SMSTEXT}</Text>
Additional Header 1-6	SMS HTTP protocols	Extra HTTP headers for SMS requests.	Authorization: Bearer fictional-token

JSON Web URL

JSON Web URL post data uses one line per JSON field in field.path:value[:type] format. Dot notation creates nested objects. Numeric path parts create numeric object keys which encode as JSON array indexes when sequential. Use :array to force a single-value array.

sender.number:${SMSSOURCENUM}
sender.name:${SMSNAME}
messages.0.destination:${SMSDESTNUM}
messages.0.content:${SMSTEXT}
message.metadata.tenant:${TENANTNAME}
recipients:${SMSDESTNUM}:array

Web URL

Web URL post data is sent as a raw form-style body. The script does not automatically URL encode values, so encode values explicitly when the provider requires it.

SOAP Web URL

SOAP Web URL sends the configured SOAP/XML body and optional SOAPAction header. Set HTTP Content Type to text/xml unless the provider documents a different value.

SMSPortal

SMSPortal uses the Client ID and API Secret to request a bearer token from the SMSPortal REST API before sending the message payload.

SIP MESSAGE

SIP MESSAGE sends the SMS through a selected SIP provider. Choose a SIP provider that is trusted and configured for message delivery.

AGI Script

AGISCRIPT runs the named AGI script for SMS delivery. Use this only for custom integrations maintained by the PBX administrator. The SMS URL/AGI Script Name field stores the script name.

Available SMS Variables

Variable	Meaning
${SMSDESTNUM}	Destination number after SMS routing and digit handling.
${SMSSOURCENUM}	Source number for the message.
${SMSNAME}	Peer or sender name available to the SMS script.
${SMSTEXT}	Message text.
${EMERGENCYNOTES}	Emergency notes from the extension when available.
${TENANTNAME}	Tenant name.
${BILLINGCODE}	Tenant billing code.
${EXTRACODE}	Tenant extra code.
${EXTRAFIELD001}, ${EXTRAFIELD002}, ${EXTRAFIELD003}	Tenant custom extra fields.