[asterisk-users] How to disable srtp in asterisk 1.8.2.3?
Miguel Baptista
miguel.baptista at uninett.no
Wed Feb 2 02:29:22 CST 2011
Hi Danny,
/srtpcapable=no/ in /sip.conf/ didn't work. Asterisk still wants to
establish a SRTP session.
Regarding your "smart" answer. I disabled the module /res_srtp.so/ in
/modules.conf /but even so asterisk still tries to establish the SRTP
session. Because the module is not loaded I get an error and the call is
establish with "normal" RTP.
/*CLI> == Using UDPTL CoS mark 5/
/ == Using SIP RTP CoS mark 5/
/[Feb 1 09:08:50] ERROR[2225]: chan_sip.c:27972 setup_srtp: No SRTP
module loaded, can't setup SRTP session./
I want to configure asterisk in a way that it doesn't even tries to
establish SRTP session ... only "normal" RTP.
Thanks for you help
- Miguel Baptista
On 01.02.2011 18:30, Danny Nicholas wrote:
>
> *From:* asterisk-users-bounces at lists.digium.com
> [mailto:asterisk-users-bounces at lists.digium.com] *On Behalf Of *Miguel
> Baptista
> *Sent:* Tuesday, February 01, 2011 11:23 AM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [asterisk-users] How to disable srtp in asterisk 1.8.2.3?
>
>
>
> Hi again,
>
> Nobody knows how to disable it? Can at least someone pinpoint me where
> can I check the latest documentation regarding SRTP. Maybe something
> might have change in the meanwhile 'Cause so far it looks like there
> is a bug in asterisk.
> Well, maybe I should report this bug then.
>
> - Miguel Baptista
>
>
> On 28.01.2011 18:22, Miguel Baptista wrote:
>
> Hi all,
>
> I upgraded one of our servers running asterisk 1.6.X to 1.8.2.3. I
> compiled it with SRTP support.
> Everything seems to work OK but I am having a weird issue. I cannot
> disable SRTP. I tried the /encryption=no/ in /sip.conf /and the
> /_SIPSRTP_CRYPTO=disable/ on my dailplan and it keeps trying to use
> the SRTP.
> Well, right now I have to have/ noload=res_srtp.so/ on my
> /modules.conf /otherwise I cannot place SIP calls (cause other ends
> don't support it)
>
> Regards,
>
> Miguel Baptista
>
>
>
> Now that my "smart" answer is out of the way, did you try
> - srtpcapable=no
> - in sip.conf?
>
> reference: http://www.voip-info.org/wiki/view/Asterisk+SRTP
>
>
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
> http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
<javascript:void(0);> <#>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110202/b9603613/attachment.htm>
More information about the asterisk-users
mailing list