Pjsip issue when using allow=all

Asterisk Asterisk SIP
1

Hi,

I’m facing an issue with PJSIP when i set the codec configuration to allow=all, for incoming calls it works without issues, with outgoing calls the asterisk sends the invite without the SDP, I’m using version 18.9.0

Thanks

2

It’s a known issue[1]. Realistically you should be explicit on codecs.

[1] [ASTERISK-29185] chan_pjsip: Endpoint: allow = all is broken. - Digium/Asterisk JIRA

3

Thanks for the confirmation! any ETA on the fix?

4

No. There is noone actively working on the issue.

1 Like
5

Specifying allow=all isn’t a sensible thing to do Even when it works, it creates requests that are likely to need to be fragmented in transit.

6

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