res_pjsip - Asterisk Documentation

Publisher

Sangoma / Asterisk project

URL

https://docs.asterisk.org/Latest_API/API_Documentation/Module_Configuration/res_pjsip/

Published

unknown

Updated

unknown

Accessed

2026-09-24

HTTP status

200

License

Asterisk documentation (project site); no-redistribution; short excerpts and locators only

Source notes citing this source

• The PJSIP endpoint direct_media option, which determines whether media may flow directly between endpoints, defaults to yes. in context
• The PJSIP endpoint context option names the dialplan context used for inbound sessions from that endpoint and defaults to default. in context
• The PJSIP endpoint media_encryption option accepts no, sdes or dtls, defaults to no, and enforces the chosen media encryption for that endpoint. in context
• In Asterisk, dtmf_mode auto sends RFC 4733 if the other side supports it and in-band otherwise; auto_info sends RFC 4733 if supported and SIP INFO otherwise. in context
• Asterisk res_pjsip endpoint dtmf_mode defaults to rfc4733 and accepts rfc4733, inband, none, info, auto and auto_info. in context
• Asterisk documentation describes rfc4733 DTMF as sent out of band of the main audio stream, and Genesys describes RTP Events as out-of-band. disputed in context
• Asterisk res_pjsip defaults timers_min_se to 90 seconds and timers_sess_expires to 1800 seconds. in context
• Asterisk res_pjsip endpoint option timers defaults to yes and accepts no, yes, required, always and forced (an alias of always). in context

Cite this source record

APA

WarmTransfer. (2026, September 24). res_pjsip - Asterisk Documentation. WarmTransfer. https://warmtransfer.net/knowledge/sources/asterisk-docs-res-pjsip-config

BibTeX

@misc{warmtransfer-asterisk-docs-res-pjsip-config,
  title  = {res_pjsip - Asterisk Documentation},
  author = {{WarmTransfer}},
  year   = {2026},
  url    = {https://warmtransfer.net/knowledge/sources/asterisk-docs-res-pjsip-config},
  note   = {Sangoma / Asterisk project, accessed 2026-09-24}
}