Hello David,
I'd recommend trying http://sipjs.com/ , it's similar to sipjs but you can
choose which kind of media it uses via a configuration object:
http://sipjs.com/guides/make-call/
Check out the guides, they are extremely clear and informative:
http://sipjs.com/guides/
cheers,
Olli
Oops, quite right, how typoful of me!
Thanks for the excellent points, I'll look into gluster and puppet and see
may way onwards from there.
cheers,
Olli
2015-02-06 12:32 GMT+02:00 Gareth Blades mailinglist+aster...@dns99.co.uk:
On 06/02/15 07:54, Olli Heiskanen wrote:
My goal is to allow
Hello,
Got a question regarding custom announcements in Asterisk.
My goal is to allow my users record their own queue announcements and
choose which announcements they want to use in each queue. I have several
Asterisk servers and a Kamailio server which dispatches call traffic
between the
Hello,
I'm stuck with getting cdr records stored in MySql database. I have a
working realtime environment and have verified that the db connection works
fine when used via res_config_mysql.conf. I'd appreciate Your help on how
to get the odbc connector working as I think there's something wrong
work with 'MySQL-asterisk' as the DSN instead of simply 'MySQL' ?
I have machines that use /etc/odbc.ini and machines that use
/usr/local/etc/odbc.ini depending on if I used a package to instal ODBC or
if I compiled ODBC myself.
On Tue, Feb 3, 2015 at 1:35 AM, Olli Heiskanen
Perfect, that's it! Thank you Paddy for pointing that out to me, I had
totally missed it!
Thanks,
Olli
2015-01-05 15:15 GMT+02:00 Paddy Grice pa...@wizaner.com:
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Olli Heiskanen
Hello all,
Just wondering on a behavior I noticed while testing with realtime sip
peers with names like 111@mydomain.com. Using Kamailio as outbound
proxy, it sends Asterisk a sip message where To header value is
sip:111@mydomain.com and From header has value username
Hello,
I'd appreciate your comments on the following problem I'm having, please
forgive me if this is something obvious, I've been scratching my head on
this for a while:
I have Asterisk+Kamailio setup where I'm currently testing inbound calls
from outside. I have both webrtc and sip clients,
2014-12-05 18:53 GMT+02:00 Gareth Blades mailinglist+aster...@dns99.co.uk:
On 05/12/14 16:46, Olli Heiskanen wrote:
INVITE that Asterisk (at port 5070) receives:
PU.BL.IC.IP:5060 PU.BL.IC.IP:5070: SIP, length: 1046
INVITE sip:6...@testers.com;transport=UDP SIP/2.0
Record-Route
about what's wrong with my setup. As my c is not
exactly fluent I wasn't sure which code files to search, can you guys help
out with that?
cheers,
Olli
2014-10-03 11:31 GMT+03:00 Matthew Jordan mjor...@digium.com:
On Thu, Oct 2, 2014 at 10:18 AM, Olli Heiskanen
ohjelmistoarkkite...@gmail.com
.
Thank you sir, I will raise a drink for you next time I'm out.
cheers,
Olli
2014-10-07 16:55 GMT+03:00 Joshua Colp jc...@digium.com:
Olli Heiskanen wrote:
Hi,
Thanks Matthew for trying to reproduce the problem, I appreciate your
efforts very much.
There must be something off in my setup
Hi,
Is there anything I can do with this problem? Re-installing Asterisk does
not solve this and the problem still persists. Or is there any other logs
or configurations I can provide to help figure out why Asterisk is removing
lines from the sdp?
Any ideas would be greatly appreciated! I also
and will **always** replace
the SDP with its own.
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Olli Heiskanen
*Sent:* Thursday, October 02, 2014 9:06 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re
Hello,
I have a problem with a call between 2 webrtc clients. Asterisk removes the
ice-related lines from the sdp when it sends the INVITE out, and the called
webrtc client rejects the INVITE due to the missing ice lines. Both webrtc
clients are defined exactly the same way, same values in all
:6...@testers.com;tag=856i7ei98p
Call-ID: oc0ppijresm05k2emsgt
CSeq: 3394 ACK
Content-Length: 0
-
--- (8 headers 0 lines) ---
u363id562*CLI
2014-09-08 17:57 GMT+03:00 Matthew Jordan mjor...@digium.com:
On Mon, Sep 8, 2014 at 9:48 AM, Olli Heiskanen
ohjelmistoarkkite
: 597260a76cb0cb9155392f3a3c0be...@testers.com
CSeq: 102 ACK
Content-Length: 0
Thanks,
Olli
2014-09-08 18:50 GMT+03:00 Matthew Jordan mjor...@digium.com:
On Mon, Sep 8, 2014 at 10:19 AM, Olli Heiskanen
ohjelmistoarkkite...@gmail.com wrote:
Hi Matthew,
Here's the debug output:
--- SIP
Hello,
I was testing with sdp and something came up worth asking:
While calling from a webrtc client to another (chrome, sip.js) Asterisk
receives the following sdp and rejects it with 488 Not Acceptable. Why does
this happen, what's wrong with the sdp? The second sdp body below is
accepted
to bridge to
RTP/AVP and RTP/AVPF only if the client cannot speak securely.
I'd very much like to hear opinions and thoughts on these.
cheers,
Olli
2014-08-13 20:39 GMT+03:00 Olli Heiskanen ohjelmistoarkkite...@gmail.com:
Aaa now I understood better, thanks!
That's the instruction I used
as srtp is on
its way anyway in future Asterisk versions and the rtp flowing between
Kamailio and users' networks are far more important than internal rtp
traffic.
cheers,
Olli
2014-08-15 18:48 GMT+03:00 Paul Belanger paul.belan...@polybeacon.com:
On Fri, Aug 15, 2014 at 10:41 AM, Olli
2014-08-12 17:40 GMT+03:00 Paul Belanger paul.belan...@polybeacon.com:
On Tue, Aug 12, 2014 at 4:17 AM, Olli Heiskanen
ohjelmistoarkkite...@gmail.com wrote:
Hello,
Thank You Paul for your reply,
The registrations in my setup are not duplicated, the 'secret' field in
the
realtime table
+03:00 Paul Belanger paul.belan...@polybeacon.com:
On Wed, Aug 13, 2014 at 4:35 AM, Olli Heiskanen
ohjelmistoarkkite...@gmail.com wrote:
Hi,
Wow, thanks Paul, realizing the problem makes a lot of sense.
So I setup Kamailio as a peer, but if I disable chan_sip module
completely,
I
:45 AM, Olli Heiskanen
ohjelmistoarkkite...@gmail.com wrote:
Hello,
I'm trying to get calls working between websocket clients and sip
clients.
For clients I have sip.js based clients on chrome, Zoipers and a
Grandstream
phone. Challenge here is I'd like to have Kamailio and rtpengine
on chrome, and calls have worked
before... I wonder if I should revert further back and/or change or remove
some realtime table fields?
cheers,
Olli
2014-08-12 11:17 GMT+03:00 Olli Heiskanen ohjelmistoarkkite...@gmail.com:
Hello,
Thank You Paul for your reply,
The registrations in my setup
Hello,
I'm trying to get calls working between websocket clients and sip clients.
For clients I have sip.js based clients on chrome, Zoipers and a
Grandstream phone. Challenge here is I'd like to have Kamailio and
rtpengine to handle the bridging between different rtp profiles but
Asterisk
to this if I have to revert back to my
previous settings.
cheers,
Olli
2014-08-05 16:49 GMT+03:00 Daniel-Constantin Mierla mico...@gmail.com:
On 01/08/14 10:56, Olli Heiskanen wrote:
Hi,
I got ahead with my setup, this post helped me much:
http://forums.digium.com/viewtopic.php?f=1t=90167sid
Hello,
I noticed a strange thing while testing my Asterisk-Kamailio Realtime
setup. In an INVITE the From and To headers contain the same number when
calling through a Realtime integration setup. This happens when the INVITE
leaves Asterisk.
Can you guys tell me what might be causing this? I
Hi,
There we go, that was it. Thank you Joshua!
cheers,
Olli
2014-08-06 15:26 GMT+03:00 Joshua Colp jc...@digium.com:
Olli Heiskanen wrote:
Hello,
Kia ora,
I noticed a strange thing while testing my Asterisk-Kamailio Realtime
setup. In an INVITE the From and To headers contain
dispatcher in Kamailio to route calls to Asterisk.
Kamailio sounds like the logical place, but I'd rather find a way to not
change the rtp profile along the way, at least until the clients can
support that one.
cheers,
Olli
2014-07-26 12:58 GMT+03:00 Olli Heiskanen ohjelmistoarkkite...@gmail.com
Greetings,
I've noticed a problem that might originate from my Asterisk configuration,
could use a hand in sorting it out. Problem is a 488 response from Asterisk
whenever it gets RTP/SAVPF profile in the SDP.
My current setup has Asterisk Kamailio realtime integration, and Kamailio
uses
Heiskanen ohjelmistoarkkite...@gmail.com:
Wow, thanks Joshua, it would've taken me forever to find the answer there.
It did the trick and the registrations look much better.
Merci beaucoup!
- Olli
2014-07-15 16:26 GMT+03:00 Joshua Colp jc...@digium.com:
Olli Heiskanen wrote:
Thanks
Hello all,
I have an Asterisk installation with Kamailio using realtime integration. I
have gotten my clients to register, but there is something odd about the
sip message flow with some of my clients. My clients are Zoiper and
Asterisk is 11.10.2.
When I set 'Subscribe to MWI' value to 'both',
this would
happen?
cheers,
Olli
2014-07-15 15:40 GMT+03:00 Joshua Colp jc...@digium.com:
Olli Heiskanen wrote:
Hello all,
Bonjour,
I have an Asterisk installation with Kamailio using realtime
integration. I have gotten my clients to register, but there is
something odd about the sip
/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb
How would I fix this double-AOR problem, can it be fixed on Asterisk
configuration?
thanks,
Olli
2014-07-15 16:00 GMT+03:00 Joshua Colp jc...@digium.com:
Olli Heiskanen wrote:
Hello,
Thanks for your response, I actually verified that the Zoiper setting
Wow, thanks Joshua, it would've taken me forever to find the answer there.
It did the trick and the registrations look much better.
Merci beaucoup!
- Olli
2014-07-15 16:26 GMT+03:00 Joshua Colp jc...@digium.com:
Olli Heiskanen wrote:
Thanks, there are no register lines in my sip.conf
answers with a Unauthorized and provide a nonce to be
used for the next registration attempt, using it to encrypt the password.
Leandro
2014-05-14 13:12 GMT+02:00 Olli Heiskanen ohjelmistoarkkite...@gmail.com
:
Hello,
After a small break from working on this, I got the idea of tcpdumping
-04-24 11:27 GMT+03:00 Olli Heiskanen ohjelmistoarkkite...@gmail.com:
Hello all,
I've been testing a Kamailio Asterisk Realtime integration, and found a
strange situation.
My problem is that when using the integration, everything seems ok but
Asterisk does not see the clients as registered
Hello,
I'm far from being an expert, but as far as I know when you use https in
your website the browser will ask to use the audio devices only once and
then remembers your decision. When using http it will ask every time.
Sorry I can't be of more help but hope this helps.
cheers,
Olli
Hello all,
I've been testing a Kamailio Asterisk Realtime integration, and found a
strange situation.
My problem is that when using the integration, everything seems ok but
Asterisk does not see the clients as registered. Kamailio and the clients
report registered clients. Also calls fail.
In
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