Greetings,
I use TLS and SRTP on all my extensions. I use openssl and distribute my root
certificate to my endpoints. Most of the time my calls work just fine.
Sometimes I receive a repeating error in my log files however, and I don’t know
why this is happening. I’m wondering if this
assigned the client device.
Does asterisk send RTP traffic to the IP which is in the IP headers of the SIP
REGISTER , or can a client “specify” it’s truly reachable IP ?
I hope this makes sense.
Regards,
Kevin Long
of other
applications, and am curious if anyone has a working example or if this is
even possible?
Thank you,
Kevin Long
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suite);
> On May 30, 2016, at 11:49 AM, Kevin Long <kevin.l...@haloprivacy.com> wrote:
>
>
>
> Hi folks,
>
>
> At least several endpoints (soft phone and desk phones) are supporting
> various 256 bit ciphers for SRTP these days. I *believe* libs
with the know-how be willing/able to submit a patch ?
Thank you,
Kevin Long
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unclear as to
whether I truly need 2 separate public IPs for the turn server to work, which I
have seen mentioned in some of the documents.
Thank you for your time.
Regards,
Kevin Long
smime.p7s
Description: S/MIM
Greetings.
I am using the PJSIP driver with TLS transport, and my endpoints are SIP mobile
apps operating in environments that I do not control.
I would like Asterisk to default to sending INVITES and all other SIP signals
to endpoints via the existing SIP TLS connection which is already
k you,
Kevin Long
output from “openssl ciphers” on my Asterisk box:
ECDHE-RSA-AES256-GCM-SHA384
ECDHE-ECDSA-AES256-GCM-SHA384
ECDHE-RSA-AES256-SHA384
ECDHE-ECDSA-AES256-SHA384
ECDHE-RSA-AES256-SHA
ECDHE-ECDSA-AES256-SHA
DHE-DSS-AES256-GCM-SHA384
DHE-RSA-AES256-GCM-SHA384
DHE-RSA-AES256-SHA256
DH
, firewall, or Asterisk/pjsip that is the culprit .
Regards,
Kevin Long
smime.p7s
Description: S/MIME cryptographic signature
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, and
wondering if anyone has anything I could try to fix or mitigate the problem in
ESXi environment .
We have freepbx (asterisk 11 chan_sip) and test environments asterisk 13.7/8
pjsip .
Thank you again,
Kevin Long
smime.p7s
Description: S/MIME cryptographic signature
.
Their provisioning system assumes that both devices will use the same SIP
extension for auth however.
Normally we would use separate extensions and a follow-me , but if there is any
way to use the same extension, I need to figure it out.
Thank you,
Kevin Long
smime.p7s
Description: S/MIME
Thanks John,
For anyone reading this using FreePBX - simply switching the default conference
app from MeetMe to ConfBridge seems to be a drastic improvement, have not
stress tested but running a conf now with no stutter on Confbrdige app.
Cheers,
Kevin Long
> On Mar 9, 2016, at 12:17
a second factor of authentication
besides the SIP secret , since in my current setup, despite using a TLS/SSL
cert for the server, the server only verifies the client by the SIP secret.
Regards,
Kevin Long
smime.p7s
Description: S/MIME cryptographic signature
Personally I am about to try asterisk on proxmox using containers since they
run code "native". I've had timing issues on conference calls (stutter) with
VMware esxi . Not sure about KVM I hope it's also better than esxi too.
Sent from my iPhone
> On Apr 6, 2016, at 9:13 AM, Markos Vakondios
again,
Kevin Long
smime.p7s
Description: S/MIME cryptographic signature
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ot working and the internal IP being sent in the
SDP from asterisk - I applied this patch to the codebase and recompiled I am
seeing the TLS “new transport” issue again , I think.
Regards,
Kevin Long
smime.p7s
Description: S/MIME cryptographic
rge Joseph <george.jos...@fairview5.com>
> wrote:
>
>
>
>> On Thu, Mar 3, 2016 at 8:25 PM, Kevin Long <kevin.l...@haloprivacy.com>
>> wrote:
>>
>> Thanks George I appreciate the info . Being able to see what codec is in
>> use for call in p
Joseph <george.jos...@fairview5.com> wrote:
>
>
>
> On Fri, Mar 4, 2016 at 1:16 AM, Kevin Long <kevin.l...@haloprivacy.com> wrote:
> Hi George the patch was from here , you wrote it I believe . I pulled
> asterisk 13 from git, apply this patch which fixed RTP issue , bu
I am having trouble with RTP and NAT :
Below is a SIP SDP invite from a remote endpoint which is trying to call
extension 420 which is the ECHO application .
As you can see, the public IP is where the request comes in from, but the SDP
contains the private, internal IP in numerous places.
Hi Joshua,
Looking at the transmitted SIP packets from Asterisk, it looks like Asterisk
is only sending it’s own internal IP (it is behind a NAT too, with proper port
forwarding) .
I did set in my transport the external_signaling_address and
external_media_address , and I have now put
Thank you for the response Joshua .
I had rtp_symmetric=yes before I wrote the email, then I set it to no,
restart asterisk, and tried to make the call from the remote endpoint again but
still tcpdump is showing me the RTP packets are being sent from Asterisk to the
private IP.
tcpdump
Hi Joshua,
This Asterisk 13 was pulled from git master branch just 2-3 days ago:
GIT-13-d1495b .
I used this very recent source code to overcome a pjsip problem (you can see my
email list post from a few days ago)
Thanks again
smime.p7s
Description: S/MIME cryptographic signature
--
!
Kevin Long
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on the phone, the call
fails.
Perhaps this is just not documented, or may not be implemented yet. Anyone
have a thought?
Thank you,.
Kevin Long
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Hello,
I am using asterisk 14.2 and PJSIP, with TLS transport.
I’m sure I’m doing something wrong here ..
In 2 distinct softphone clients (Bria and Groundwire), I am able to register
successfully, and place a SIP call, with no certificate warnings. But shortly
after I place that first
Hello,
All my asterisk systems use only IPv4 currently. I have one phone which is on
T-Mobile network, and this network is only IPv6 now.
The phone can register fine, because T-Mobile does NAT64 and it connects fine
to my IPv4 asterisk server.
But in the SDP for a call setup, this
Greetings !
My goal is to get Twilio trunking working, and with TLS/SRTP.
I see this concerning message in my log:
[Feb 7 16:50:26] ERROR[20596] res_sorcery_config.c: Could not create an object
of type 'endpoint' with id ’twilio' from configuration file ‘pjsip.conf’
Thus, ‘pjsip show
something like this.
Outbound is the easy part. How are you handling inbound SMS->SIP ?
Regards,
Kevin Long
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Greetings,
I am getting the following error (below) continually in my asterisk log,
related to qualify_frequency I believe. I am trying to use sip trunking with
the company flowroute.
3 questions if I may:
1) Is using qualify_frequency with a sip trunk a common or recommended
practice? I
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