How to set transport through ARI for outgoing calls

Using ARI to originate calls. Is there way to provide desired transport protocol for it without peer configuration? If I pass in just SIP/exten@host call goes out but protocol is defaulted to UDP. Can I pass desired protocol with ARI request somehow?

Asterisk 16.2.1, chan_sip

The Dial string is the same as is used when dialing from the dialplan. These are documented in the sample configuration file for chan_sip[1].

[1] asterisk/sip.conf.sample at master · asterisk/asterisk · GitHub

Thank you for quick response, that worked.

Another question though.
If I select TLS transport how can I also make asterisk to use SRTP for that call without peer configuration.
Global “encryption=yes” doesn’t seems to be doing it. Is there any param for media encryption I can pass in with ARI call origination request?

Most of my chan_sip knowledge is gone, if it’s not specified in the sample configuration file I do not know. It may not be possible.

+1 (Plus another xx characters to get past the forum threshold.)

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