Lefteris,
Thanks a lot for your detailed answer and for the valuable work you've been
doing on this topic for quite some time now.
Cheers,
Philippe Sultan
2015-08-28 12:26 GMT+02:00 Lefteris Zafiris zaf@gmail.com:
On Fri, 28 Aug 2015 12:11:14 +0300
Amelye Chatila amec...@gmail.com wrote
Hi,
I'm running an Asterisk server connected to a carrier over 2 E1 cards. From
time to time, the Called Number Party presented by the carrier changes a
bit (for some reason I don't know) and is prefixed with a byte string (e.g.
: 00 34 34 39 ), which furtherly prevents libpri from getting the
:29 PM, giovanni.v i...@keybits.org wrote:
Il 02/11/2011 15.06, Philippe Sultan ha scritto:
PRI Span: 1 [70 13 a1 00 34 34 39 39 30 30 32 30 33 36 31 35 38 39 34
32 35]
Yes, like you guessed the third bit (wich is part of the called number
i.e.) is a NUL... but Q.931 allows any IA5 (ISO
Issue filed : https://issues.asterisk.org/jira/browse/PRI-128
Philippe
On Wed, Nov 2, 2011 at 7:00 PM, giovanni.v i...@keybits.org wrote:
On 02/11/2011 17.52, Philippe Sultan wrote:
The 's' extension would match any
number, and I would not be able to retrieve the actually dialed number
:
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And by the way, app_confbridge is included in the 1.6.2 series (at least).
On Mon, Feb 8, 2010 at 1:49 PM, Philippe Sultan
philippe.sul...@gmail.com wrote:
Hi Klaus,
The module is app_confbridge, and the application is ConfBridge. I had
been using it for a while because it's really easy
Philippe, what exactly is a playback channel? Is it a pseudo participant
playing back the announcements?
Yes. Announcements are played to the conference members by creating a
channel of type 'Bridge' which streams the sound files.
thanks
klaus
Further, is there somewhere a documentation
Dear community members,
I'm happy to announce that we now have code that allows you to use
your XMPP (Jabber) client like a softphone to place SIP or PSTN (or
whatever channel Asterisk supports) calls.
The XMPP clients that support Jingle that I and others have tested are :
- Pidgin (Linux,
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Hi everybody,
From an Asterisk console, I'd like to retrieve information from SIP
users (eg. their contact address) that are registered on a Kamailio
(OpenSER) server.
Kamailio is defined as a peer in my sip.conf file, and it looks like
the 'sip qualify peer' command can help me get the
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Hi Julien,
The Gtalk call to your buddy fails because of a mismatch in the UDP
ports for RTP. Try to disable the 'strictrtp' option in your rtp.conf
file. Question : did you scramble the IP addresses?
The Jingle call fails because of Google's XMPP network refusing to
relay jingle packets wrapped
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Hi Julien,
On Sun, Oct 26, 2008 at 4:51 PM, Julien Claassen [EMAIL PROTECTED] wrote:
Hi!
There's something strange. I have entered a couple of buddies. On has Jingle
capability and two have resources (Home and Telepathy), but my own account
does have no resource, I put myself in the buddies
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Hi Julien,
bach [Oct 25 21:18:11] ERROR[28847]: chan_gtalk.c:908
gtalk_alloc: no gtalk capable clients to talk to.
[Oct 25 21:18:11] NOTICE[28847]: channel.c:3243 __ast_request_and_dial: Unable
to request channel gtalk/gtalk_account/[EMAIL PROTECTED]
The syntax is correct. Make sure that
Hi Bilal,
On Tue, Sep 23, 2008 at 11:11 PM, bilal ghayyad [EMAIL PROTECTED] wrote:
Dear Philippe;
Thanks a lot for ur kindly answer.
How can I use the Radius with CDR (Accounting)?
Here is the documentation :
http://svn.digium.com/view/asterisk/branches/1.4/doc/radius.txt?view=markup
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asterisk
at a testing stage, if you want to
give a try, please check : http://bugs.digium.com/view.php?id=12569
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Hi Julian,
How difficult would it be to have a JabberReceive Event *initiate* a
channel ?
I think that could be done. And you could also place Originate
commands over AMI, as you mentioned it. You might be interested in
BJ's work, as it covers that topic :
http://www.asterisk.org/node/48440
Hi Julian,
[...]
What can you do with it? Well, a direct usage of this application is
to make an easy to use GoogleTalk voice gateway out of Asterisk. Here
is an example (assuming the asterisk-xmpp account is configured) :
context gtalk-in {
s = {
NoOp(Caller id :
Friends,
a new dialplan application is now available for testing :
http://svn.digium.com/view/asterisk/team/phsultan/jabberreceive/
The corresponding feature request is located here :
http://bugs.digium.com/view.php?id=12569
What can you do with it? Well, a direct usage of this application is
Thanks for the snippet, I re-wrote it (badly) for regular extensions.conf
usage, and verified it's also working here on 1.6, though I do get a warning
about JabberStatus being depreciated.
Yes, JabberStatus is being moved from an dialplan application to a
function (JABBER_STATUS), because it's
Hi Benoit,
Anyone already did that (changing jabber status/ status message of many
accounts)
or know if it's even remotly possible ??
We discussed that during the last XSF devcon in Brussels. Actually
Asterisk (or any other XMPP client) cannot change the Jabber status on
behalf of another
Hi Matt,
On Wed, Jun 4, 2008 at 1:05 AM, Matthew Gibson [EMAIL PROTECTED] wrote:
I'd be interested to know more about the status abilities as well, we've
tried to test jabberstatus application, but it doesn't seem to function as
we expect, it should be returning 0,1,2,3,4,5 based on users
mailing list
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Hi Ali,
I have sent a previous email with a problem that I solved by using component
mode. In this mode the asterisk server acts as a subdomain. So I can call
[EMAIL PROTECTED], [EMAIL PROTECTED]
That's a nice way of using Asterisk's component capability. Which
XMPP/Jingle client are you
Hi Ali,
On Fri, Mar 28, 2008 at 5:31 PM, Ali Jawad [EMAIL PROTECTED] wrote:
Hi All
I am developing a client that uses libjingle to do xmpp stuff with
ejabberd. I can also make audio calls between those clients. What I am
trying to archive now is to send calls to pstn using jingle. I was
On Mon, Mar 31, 2008 at 4:51 PM, Ali Jawad [EMAIL PROTECTED] wrote:
So should I register directly on the asterisk server or should I send
the voice calls through ejabberd to asterisk ?
You can't register an XMPP client on Asterisk, because it's not an
XMPP server. The required steps to
://www.voip-info.org/wiki/view/Asterisk+Google+Talk
However, feel free to open a bug report if you've made sure you have a
properly installed iksemel stack.
Note that as of Asterisk 1.6, GnuTLS was replaced by OpenSSL which is
now used by Asterisk as the encrypting protocol for iksemel.
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I'm taking the liberty to announce this event on the Asterisk mailing
list, as Asterisk and OpenSER form a valuable combination in SIP
architectures.
The second edition of OpenSER Summit will take place in San Jose, USA
,on the 17th of March, 2008, during VonX Spring 2008 pre-conference
events.
Hi Clive,
Hi all,
Do some one experiencing running jabber applications (jabberstatus...) in
asterisk? I do experinced Asterisk 1.4.18 and wish to start it, however I
got such result.
IBM*CLI help jabber
No such command 'jabber'.
IBM*CLI help jabberstatus
No such command 'jabberstatus'.
Hi Adam,
I've been googling for half an hour, i found some sort of jingle
protocol which i'm not sure what to use for but it might be the
solution? It seems to me that my problem is sending the dtmf tones, not
receiving them, so this is really gtalk related.
You've spotted the problem,
Hi Olivier,
At the opposite, I think it could be useful for an Asterisk server to act as
XMPP User Activity provider (ie update XEP-0108 field with on-the-phone
value).
Do you agree ?
This is indeed a direction we should consider in order to relay call
and device state information to XMPP
?
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Hi Matt,
Can an Asterisk server hold logins for multiple Japper accounts on a
remote Jabber server, and carry multiple Jabber calls simultaneously the
way it can carry multiple SIP (or IAX, or ZAP, etc) calls? If so, is
each of those Jabber calls as lightweight as, say, each SIP call?
Hi Abel,
Is there a way to catch de gtalkID of a caller that´s calling my
asterisk gtalk account?
the caller id is not properly set, only its ANI part is. I just
proposed a patch in order to retrieve the CALLERID(name) variable from
the Dialplan, see http://bugs.digium.com/view.php?id=11549.
Hi Eric,
I'm looking for a SIP to XMPP Jingle voice gateway.
I see that Asterisk has Jabber and Jingle support, but it looks like
Asterisk acts as a Jabber client.
Asterisk can connect as a client or component to a XMPP server. XMPP
components are typically used as gateways between XMPP
If I calling asterisk with GTALK in english everything is ok, however, some
of my friends with the italian version of gtalk they cannot have the audio.
Audio problems might be experienced with older Gtalk clients. Version
1.0.0.104 is reported to work.
The following resources may help you :
Hi Alejandro,
the Jabber module in Asterisk is available starting from the 1.4
series. Therefore, you can connect Asterisk as a client (or component)
to your Jabber server after you've upgraded to 1.4.
You'll get detailed information here :
http://www.voip-info.org/wiki-Asterisk+Jabber
On 8/31/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Ows, I suppose that * can only do c2s to google talk to which I did and I got
audio both
ways. Yet I have not seen anything so far how * could do a s2s to google talk.
Indeed, the Jabber module was not designed to make Asterisk a Jabber
Hi Bruce,
[EMAIL PROTECTED]
Google's server is expecting you to provide a valid gmail address
here, suffixed with @gmail.com
Cheers,
Philippe
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Hi Demuel,
On 6/22/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Yeah, just the same as the sample configuration by mog. However, if I am
using a gtalk
application in asterisk to dial googletalk buddy, my voip phone is suddenly
connected to
the googletalk buddy though at the googletalk
What is the main distinction between Jingle and Gtalk here? How should I
generate the
file streamed to the SIP phone by Asterisk?
I really have no clue :). Maybe you can open a bug report so that we
can dig into this problem.
Thanks!
Philippe
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Hi Koen
This works fine when I call this account from my personal gtalk. But others
have some very strange problems.
In most cases, I see the call coming into Asterisk and executing normally.
On the callers side, the call looks like it was answered, but there's no
audio.
In some other
Philippe, what part of the channel code handles the ringing and dialling.
From my
experience here, making a call from googletalk to a voip phone inside a
firewalled
environment does not pose any problem. But making call from voip phone to
googletalk is
kinda tricky.
Well, chan_gtalk
Hi everybody,
I'd like to have the feedback from the community regarding this patch
: http://bugs.digium.com/view.php?id=9972
res_jabber currently relies on the iksemel API to handle TLS
connections, which assumes GnuTLS to be installed on the system. The
basic idea of the proposed
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