hello,
i'm searching second BOUNTY donor ($250)
for
https://issues.asterisk.org/jira/browse/ASTERISK-22708
if you want participate, please contact me privately
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Marek Cervenka
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I've been having some issues with srtp. so I checked which version of
libsrtp I built asterisk 11.6 against. I'm on fedora 21, so
libsrtp-1.4.4-13.20101004cvs.fc21.x86_64.
From https://github.com/cisco/libsrtp it seems that latest release is
1.5.1, released a couple of weeks ago.
I'm not a
I've been having some issues with srtp. so I checked which version of libsrtp I built asterisk
11.6 against. I'm on fedora 21, so libsrtp-1.4.4-13.20101004cvs.fc21.x86_64.
From https://github.com/cisco/libsrtp it seems that latest release is 1.5.1, released a couple
of weeks ago.
I'm not a
CentOS-6.5 (FreePBX-2.6)
Asterisk-11.14.2 (FreePBX)
snom870-SIP 8.7.3.25.5
I am having a very difficult time attempting to get TLS and SRTP
working with Asterisk and anything else. At the moment I am trying to
get TLS functioning with our Snom870 desk-sets. And I am not having
much luck.
Since
On 2015-03-02 22:53, ricky gutierrez wrote:
Hi list , I have a question with account codes, all my outgoing calls
are authenticated, but now the boss wants to monitor these calls with
the codes.
example: maria has an extension 110, but peter was in place and use
the phone maria , maria then
JBB == James B Byrne byrn...@harte-lyne.ca writes:
JBB tcpenable=yes
JBB tlsenable=yes
JBB tlscertfile=/etc/pki/asterisk/ca.harte-lyne.hamilton.asterisk.crt
JBB tlscafile=/etc/pki/tls/certs/ca-bundle.crt
JBB tlsdontverifyserver=yes
JBB tlscipher=ALL
JBB tlsclientmethod=tlsv1
You are missing the
These are the sip settings on our installion.
Global Settings:
UDP Bindaddress:0.0.0.0:5060
TCP SIP Bindaddress:0.0.0.0:5060
TLS SIP Bindaddress:(null)
Videosupport: No
Textsupport:No
Ignore SDP sess. ver.: No
AutoCreate Peer:
Am 03.03.2015 um 18:16 schrieb James B. Byrne:
CentOS-6.5 (FreePBX-2.6)
Asterisk-11.14.2 (FreePBX)
snom870-SIP 8.7.3.25.5
I am having a very difficult time attempting to get TLS and SRTP
working with Asterisk and anything else. At the moment I am trying to
get TLS functioning with our Snom870
On Tue, March 3, 2015 13:19, jg wrote:
Forget about the reverse DNS stuff for the moment.
Do simple SIP accounts (without SRTP/SRTP and deny/permit stuff) work?
Enable SRTP, but you likely need the AES-80 fro SRTP Auth-tag.
Then try the rest.
jg
The Snom870s and our Asterisk FreePBX
I'd like to dial two extensions (or external number) and ask for
confirmation to accept the call.
Dialing an extension, asking for confirmation and then dialing a second
extension if the call has not been accepted is easy by using the dial
option U(...), but if I dial two extensions at once, when
Other things to consider:
The transport config, which can be in [general] or in a peer's [] block.
if you want tls-only, use transport=tls
it also accepts tcp, udp or a comma-separated list.
if given a list, it tries them in order
If you need ast to register over tls, use something
On Tue, Mar 3, 2015 at 3:35 PM, Leandro Dardini ldard...@gmail.com wrote:
I'd like to dial two extensions (or external number) and ask for
confirmation to accept the call.
Dialing an extension, asking for confirmation and then dialing a second
extension if the call has not been accepted is
I reconfigured sip.conf to have these settings:
tcpenable=yes
tlsenable=yes
tlscertfile=/etc/pki/asterisk/ca.harte-lyne.hamilton.asterisk.pem
tlscafile=/etc/pki/tls/certs/ca-bundle.crt
tlsdontverifyserver=yes
tlscipher=ALL
tlsclientmethod=tlsv1
On Tue, March 3, 2015 16:34, James Cloos wrote:
Other things to consider:
The transport config, which can be in [general] or in a peer's []
block.
if you want tls-only, use transport=tls
it also accepts tcp, udp or a comma-separated list.
if given a list, it tries them in order
On Tue, March 3, 2015 13:37, James Cloos wrote:
JBB == James B Byrne byrn...@harte-lyne.ca writes:
JBB tcpenable=yes
JBB tlsenable=yes
JBB tlscertfile=/etc/pki/asterisk/ca.harte-lyne.hamilton.asterisk.crt
JBB tlscafile=/etc/pki/tls/certs/ca-bundle.crt
JBB tlsdontverifyserver=yes
JBB
Thinking about it I don't think you want to do what you are asking, It
sounds to me like you would create a race condition.
Otherwise what happens when the Person A answers and accepts the call and
Person B also answers and accepts the call?
Which channel do you bridge your call with? Person A
by removed line ca_list_file=/pbx/keys/ca.key
ERROR[3301]: pjsip:0 ?: ssl0x7fc8e40f8 Error loading CA list file
'/pbx/keys/ca.key
gone.
But still cannot handle SRTP, phone says 488 error if I set
media_encryption=sdes on an endpoint,
how do I check if srtp actually work on asterisk?
On 03
For those that were interested I have attached the kamailio.cfg which we
have working with Kamailio 4.2.1 and Asterisk 1.8.23/32. Specifically, the
following yum packages:
kamailio.x86_64 4.2.1-4.1
@home_kamailio_v4.2.x-rpms
kamailio-auth-ephemeral.x86_64 4.2.1-4.1
Hey guys,tried to make tls work with pjsip on asterisk 13.2.0
have compiled pjsip with ssl,
added transport
[tls]
type=transport
cert_file=/pbx/keys/server.crt
ca_list_file=/pbx/keys/ca.key
priv_key_file=/pbx/keys/server.key
protocol=tls
bind=192.168.1.4:5061
local_net=192.168.1.0/24
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