Hi
I want to test a SS7 connection with 3 virtual machines running with Wmware
workstation player.
I tried all configurations in network setting in Wmware but although I can ping
other machines and then see their
mac address but it seems TDMoE cannot find other side.
appreciate for any comment.
I'd say using s3fs (or similar) is an approach, but if VoiceMail had
support baked into it for S3, then the integration would be better.
I'll look into using one the FUSE based approaches as a stop-gap
measure. ;)
On Tue, 2016-02-16 at 13:12 +0100, Olivier wrote:
> Isn't the purpose of s3fs-like
OK, I fixed it. Here's what I did:
Well, I first saw a lot of errors like this when Asterisk starts up (CLI
messages immediately upon startup):
[Feb 18 22:47:44] ERROR[5749]: netsock2.c:305 ast_sockaddr_resolve:
getaddrinfo("sillyapp.pstn.twilio.com;transport=tcp", "(null)", ...): Name
or servi
On Thu, Feb 18, 2016 at 8:20 PM, Sonny Rajagopalan <
sonny.rajagopa...@gmail.com> wrote:
> Thanks George, for your mighty quick response.
>
> I made the changes (re: server_uri_pattern etc.) and still, no luck--it
> fails for the same error.
>
> BTW, there is nothing for transport (but this is the
Thanks George, for your mighty quick response.
I made the changes (re: server_uri_pattern etc.) and still, no luck--it
fails for the same error.
BTW, there is nothing for transport (but this is the same config from my
SIP/UDP + Twilio days, which worked):
*CLI> pjsip show transport twilio-siptru
On Thu, Feb 18, 2016 at 7:25 PM, Sonny Rajagopalan <
sonny.rajagopa...@gmail.com> wrote:
> Hello,
>
> I have an Asterisk 13.6.0 PBX using PJSIP connected to the Twilio gateway.
> I am able to make calls outbound through the gateway, but I am not able to
> make calls into the PBX from external PSTN
Hello,
I have an Asterisk 13.6.0 PBX using PJSIP connected to the Twilio gateway.
I am able to make calls outbound through the gateway, but I am not able to
make calls into the PBX from external PSTN.
Specifically, an incoming call is _received_ by Asterisk, but it is not
able to route the call i
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Le 18/02/2016 11:03, Richard Mudgett a écrit :
> I've been using Grandstream phones for more than 10 years, but onl
y
> yesterday tried to use Early Dial... and I failed. What is needed
on the
> Asterisk side to reply 484 to INVITE? Phones
Tonight community services may have intermittent availability due to
maintenance. This maintenance will begin at approximately 8:00 PM
CST[1] and should last no longer than three(3) hours, ending around
11:00 PM CST.
The affected services are:
* All Asterisk community services including everythin
That makes sense, so its not possible to have option 'tT' in DIAL() and
have directmedia at the same time.
Thanks Richard,
Regards,
Sammy
On Thu, Feb 18, 2016 at 4:42 PM, Richard Mudgett
wrote:
>
>
> On Thu, Feb 18, 2016 at 3:05 PM, SamyGo wrote:
>
>> Hi All,
>> I've been wondering if I can i
What about leaving the old PBX in place and trunking it via ISDN to the
asterisk server.
We use rhino 24 channel bank but are 2U for rhino + 1U for patch panel.
(RJ21 cable so might be able to use existing ones if they are RJ21)
Used USB xorcoms a while back, things may of changed but if one is d
On Thu, Feb 18, 2016 at 3:05 PM, SamyGo wrote:
> Hi All,
> I've been wondering if I can instruct asterisk in the dialplan to engage
> the Media handling for a particular call or not.
>
> I've SIP users behind Kamailio & RTPProxy, and I can make use of sip.conf
> setting "directmediadeny|directmed
Hi All,
I've been wondering if I can instruct asterisk in the dialplan to engage
the Media handling for a particular call or not.
I've SIP users behind Kamailio & RTPProxy, and I can make use of sip.conf
setting "directmediadeny|directmediapermit" to offload media from asterisk
for peer-to-peer ca
On Thu, Feb 18, 2016 at 2:42 PM, Jean-Denis Girard
wrote:
> -BEGIN PGP SIGNED MESSAGE-
> Hash: SHA1
>
> Hi list,
>
> I've been using Grandstream phones for more than 10 years, but only
> yesterday tried to use Early Dial... and I failed. What is needed on the
> Asterisk side to reply 484
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi list,
I've been using Grandstream phones for more than 10 years, but only
yesterday tried to use Early Dial... and I failed. What is needed on the
Asterisk side to reply 484 to INVITE? Phones are talking to chan_pjsip
on Asterisk-13.7.1.
Thanks,
On 2016-02-17 16:28, Richard Mudgett wrote:
On Wed, Feb 17, 2016 at 5:56 PM, Ernie Dunbar
wrote:
On 2016-02-17 15:32, Richard Mudgett wrote:
On Wed, Feb 17, 2016 at 5:15 PM, Ernie Dunbar
wrote:
Hi everyone.
We have an Asterisk server running Debian Squeeze, with Asterisk
v1.8.13.1 (basicall
On Thu, Feb 18, 2016 at 4:29 AM, Olivier wrote:
> Hello,
>
> I'm having my first steps with WebRTC.
>
> I've found this line in http.conf.sample (asterisk 13.7.0):
> ;tlsprivatekey=; path to private key file
> (*.pem) only.
>
>
> Is it a typo ?
>
Not really. The private key file must be in
Sonny Rajagopalan wrote:
George,
May I propose we improve the documentation on the Asterisk Wiki? I
thought I would have spent far less time here (though you folks have
been mightily helpful, and thanks again!) should the documentation for
the TCP transport be improved in both the Wiki and speci
George,
May I propose we improve the documentation on the Asterisk Wiki? I thought
I would have spent far less time here (though you folks have been mightily
helpful, and thanks again!) should the documentation for the TCP transport
be improved in both the Wiki and specifically, in
${ASTERISK_HOME
2016-02-18 15:42 GMT+01:00 Marek Červenka :
> my experience with pjsip for webrtc
> http://lists.digium.com/pipermail/asterisk-users/2015-September/287562.html
>
>
> Yes I saw this post earlier today.
Having to fight 14 days scared me a bit !
Did you set sipml5 on your own server or did you use L
Hi,
All of the back and forth of Analog vs VoIP handsets tend to ignore some
of the basic issues.
What type of cabling is in place, does it need to be upgraded to do
Ethernet, etc.
The "gateways" can be handy if there are many closets where the wiring
terminates AND you have Ethernet to th
my experience with pjsip for webrtc
http://lists.digium.com/pipermail/asterisk-users/2015-September/287562.html
Dne 18.2.2016 v 15:36 Olivier napsal(a):
2016-02-18 14:57 GMT+01:00 Simon Hohberg
mailto:simon.hohb...@mcs-datalabs.com>>:
Is it implied here that both HTTPS and WSS mus
2016-02-18 14:57 GMT+01:00 Simon Hohberg :
>
> Is it implied here that both HTTPS and WSS must also come from the same
>> server (Same Origin Policy) ?
>>
> No, the same origin policy does not apply to web sockets.
>
> Then, can I also install my own WebRTC demo page on my own private
>> Asterisk
Is it implied here that both HTTPS and WSS must also come from the
same server (Same Origin Policy) ?
No, the same origin policy does not apply to web sockets.
Then, can I also install my own WebRTC demo page on my own private
Asterisk server and access this demo page through HTTPS ?
If I'm
Thank you much for yor reply.
2016-02-18 13:30 GMT+01:00 Simon Hohberg :
> Hi Oliver,
>
> On 02/18/2016 12:10 PM, Olivier wrote:
>
> Hello,
>
> I'm trying to have my first calls with WebRTC.
> My server has asterisk 13.7.0.
>
> I'm following the instructions from the wiki [1].
> So I'm using [2]
Hi
We are using asterisk 1.8.23.1 on CentOS 6
Is there a way that transferring by SIP REFER can be blocked on a call by
call basis?
Regards
Ish
--
Ishfaq Malik
Department: VOIP Support
Company: Packnet Limited
t: +44 (0)161 660 2350
f: +44 (0)161 660 9825
e: i...@pack-net.co.uk
w: http://www
Hi Oliver,
On 02/18/2016 12:10 PM, Olivier wrote:
Hello,
I'm trying to have my first calls with WebRTC.
My server has asterisk 13.7.0.
I'm following the instructions from the wiki [1].
So I'm using [2] live demo from a Chrome navigator (v48) on Debian
Jessie station.
Whenever I type somethi
Hello,
I'm having my first steps with WebRTC.
I've found this line in http.conf.sample (asterisk 13.7.0):
;tlsprivatekey=; path to private key file (*.pem)
only.
Is it a typo ?
I expected something like:
;tlsprivatekey=; path to private key file (*.key)
only.
Regards
--
__
Hello,
I'm trying to have my first calls with WebRTC.
My server has asterisk 13.7.0.
I'm following the instructions from the wiki [1].
So I'm using [2] live demo from a Chrome navigator (v48) on Debian Jessie
station.
Whenever I type something like ws://123.123.123.123:8088/ws in Expert Mode
for
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