Asterisk Project Security Advisory - AST-2017-014
ProductAsterisk
SummaryCrash in PJSIP resource when missing a contact
header
The Asterisk Development Team would like to announce security releases for
Asterisk 13, 14 and 15, and Certified Asterisk 13.18. The available releases are
released as versions 13.18.5, 14.7.5, 15.1.5 and 13.18-cert2.
These releases are available for immediate download at
Hi,
do you have access to the system that sends you these calls?
If it's also an Asterisk, you could tell it to send another INVITE URI,
regardless of what is submitted
in the registration.
On Asterisk with chan_sip you can do it by dialling:
On Fri, Dec 22, 2017, at 9:54 AM, Benoit Panizzon wrote:
> Dear List
>
> It looks like the common way to to sip signaling over a trunk is:
>
> In the Request URI, return the 'Register' Contact.
> In the To: Header, send the destination number.
>
> Unfortunately, asterisk with pjsip (i did not
Dear List
It looks like the common way to to sip signaling over a trunk is:
In the Request URI, return the 'Register' Contact.
In the To: Header, send the destination number.
Unfortunately, asterisk with pjsip (i did not try chan_sip) does
expect the dialed extension as request uri and does
