The java api is 99% the same as the python/lua/perl api. You can
check the wiki and the sample scripts in the source tree. This may be
incomplete, feel free to ask any questions here where you can't find
samples and we can try to fill in the missing pieces if they are not
on the wiki.
M
Hi,
I have managed to hook my Java program in. Now, looking for some hint, how
we can control the call. Actually, I need to make a call, on genuine mobile
phone, using Gizmo5 VoIP network. Furthermore, from where I can get the
Mod-Java API documentation.
--
Best,
Adeel Ansari
http://www.linkedi
It worked, Anthony. Thank you for the help.
Cheers.
On Mon, Aug 11, 2008 at 10:31 PM, Anthony Minessale <
[EMAIL PROTECTED]> wrote:
> This is already fixed in the latest SVN.
>
> The mod_java was using a older method of initialization and we forgot to
> bring it up to date.
>
>
>
> On Mon, Aug 11
Just testing one I had 23 active calls in a conference ( half SIP and half
PSTN ). This was with real people ( desk phones and cell phones ).
Brian S/bsnipes
On Monday 11 August 2008 10:30:04 am Fernando Testa wrote:
> Hi list,
>
> I'm impressed by the quality of FS and I'm considering to sugge
I am coming in a little late on this conversation but you can have both with
keeping the existing mailman system. I added freeswitch-users to Nabble
groups back in March. The same thing can be done to freeswitch-dev and we
might be able to import the even older freeswitch-users data.
For thos
To all those concerned…and I know there were many of you, my problem is solved.
I started from scratch basically…
Because I am just learnt many things…obviously while I was still learning I
must have changed a setting that caused the IVR to not work properly…
Having said that I don’t know wh
Maybe I should read first ?
http://adhearsion.pbwiki.com/FreeSwitchHelloWorld
Anyhow - opinions ?
> Hi FS community,
> I remember, that GUI issue, and actually many other FS control access
> issues are of high priority, and seems there is something what
> might/should help:
> http://adhearsion.c
I remove retun error on mod_sndfile and recompile it's work for me.
Now i'm testing other feature. i think FS is excellent :)
I will add Thai lang support and send back to svn. but may be only me
use FS in Thailand :)
Dome C.
On Tue, Aug 12, 2008 at 3:41 AM, Anthony Minessale
<[EMAIL PROTECTED
Hi FS community,
I remember, that GUI issue, and actually many other FS control access
issues are of high priority, and seems there is something what
might/should help:
http://adhearsion.com/
It is not straitforward since it was made with Asterisk in mind, but I
believe it shouldn't be big issue
Here's the page for this week's documentation call. As usual it is
currently thin, but I'm sure there are things people can add. We'll be
picking a new "mod of the week" so if you have an opinion then please
hop on the call.
http://wiki.freeswitch.org/wiki/Wiki_meet_2008_08_13
Thanks,
fixed in tree, try now
On Sun, Aug 10, 2008 at 11:19 PM, Michael Jerris <[EMAIL PROTECTED]> wrote:
> We discussed this a while back and could not come up with any
> compelling use cases where this should happen with the exception of
> configuration error. Do you have one?
>
> Mike
>
> On Aug 10,
Ruchir,
I have been using the event socket with good success. It's too much to
discuss right here so I will try and start a wiki page on it.
-MC
Sent from my iPhone
On Aug 10, 2008, at 2:03 PM, Ruchir Brahmbhatt <[EMAIL PROTECTED]
> wrote:
> Hi,
>
> Which is the best method of doing call b
I have a MV370 GSM gateway from Portech using in *.
The extension number of the gateway is 2001
To dial via the gateway I use a prefix 9.
;GSM VIA MV-370 GATEWAY)
exten => _9X.,1,Dial(SIP/2001/${EXTEN:1})
exten => _9X.,n,Congestion()
This works well.
Now I'm in the process of switching that gat
We tend not to get involved in performance number related questions but
maybe some users can share their numbers.
On Mon, Aug 11, 2008 at 10:30 AM, Fernando Testa <
[EMAIL PROTECTED]> wrote:
> Hi list,
>
> I'm impressed by the quality of FS and I'm considering to suggest our
> company to use it
The user you are calling is probably not registered.
On Mon, Aug 11, 2008 at 2:14 AM, Adeel Ansari <[EMAIL PROTECTED]> wrote:
> Hi Guys,
>
> I am having this error on Freeswitch console. Any ideas.
>
> *[ERR] switch_ivr_originate.c:926 switch_ivr_originate() Cannot create
> outgoing channel of ty
you mean sofia-sip-devel list ?
Cheers
Kirk
2008/8/11 Michael Jerris <[EMAIL PROTECTED]>
> Try posting this to the sofia-sip mailing list, lets see if we can sort out
> a fix there.
> Mike
>
> On Aug 11, 2008, at 11:25 AM, Kirk Bateman wrote:
>
> Michael,
>
> Great :( so probably something I'd
Try posting this to the sofia-sip mailing list, lets see if we can
sort out a fix there.
Mike
On Aug 11, 2008, at 11:25 AM, Kirk Bateman wrote:
Michael,
Great :( so probably something I'd need to mod the sofia source for
really then.
Not much chance of me getting Tesco (technically its
Hi list,
I'm impressed by the quality of FS and I'm considering to suggest our
company to use it for the conference platform. So, since I don't have
currently a way to stress it, the concerns are:
Q1: how is the quality under load?
Q2 what are the limits of a conference bridge? Usually we get from
Michael,
Great :( so probably something I'd need to mod the sofia source for really
then.
Not much chance of me getting Tesco (technically its a rebadge of freshtel /
voicedot) to change their server so its compliant :)
Cheers
Kirk
2008/8/11 Michael Jerris <[EMAIL PROTECTED]>
> The issue fro
The issue from what I can see in the trace is the start of the s and o
lines. We saw this before in a slightly different variant where those
lines had extra whitespace in them after the =, this is probably the
same thing, illegal chars after the =.
Mike
On Aug 11, 2008, at 11:09 AM, Kirk B
Anthony,
Yes I'd figured that out :)
I couldn't see anything specific that it was complaining about ... I've
looked at the source and not figured it out yet... (really must try and
memorize the spec someday).
I was wondering if it was something to do with the X-NSE bit (dtmf tones
extension to r
I assume it's "whining" about the SDP parse error encountered.
nua(0x8120fe0): INVITE server: error parsing SDP
I do not have the spec memorized but I'm sure if we show it to the sofia dev
he can tell us the specific problem with the sdp.
On Mon, Aug 11, 2008 at 8:30 AM, Kirk Bateman <[EMAIL P
This is already fixed in the latest SVN.
The mod_java was using a older method of initialization and we forgot to
bring it up to date.
On Mon, Aug 11, 2008 at 3:26 AM, Adeel Ansari <[EMAIL PROTECTED]> wrote:
> Below is the error, I am getting, while running my Java program through
> Freeswitch
A few weeks ago the voip-info dude emailed me letting me know he made this
one if we want to use it.
http://www.voip-info.org/boards/index.php?b=6
*shrug* we are already almost forgetting about it but you are welcome to
try and wake it up.
On Mon, Aug 11, 2008 at 4:59 AM, Jonas Gauffin <[EMA
Afternoon everyone,
I have a bit of a problem with Freeswitch receiving RTP INVITEs from my SIP
provider (tesco's internet phone ... I know SIP isn't supported technically,
but it sort of works...)
Freeswitch (sofia) seems to be whinging about the INVITE SDP format ... but
I'm not sure why ... he
Why? Do you actually think that a FreeSWITCH forum would become a
warzone? Isn't the userbase a bit too grown up for that?
Google groups sounds like an excellent compromise.
Theres a google group for google groups:
http://groups.google.com/group/Google-Groups-Guide
And here is a guy that wrote an
Brian West wrote:
> The only worry about forums is that it splits support resources
> between the two. Any thoughts?
>
Yeah, I have a thought. Would anyone who thinks forums are a good idea
care to name one that actually works, isn't a war zone, and yet
functions without truly massive full
X-ECN Telecoms-MailScanner-Information: Contact ECN Telecoms
X-ECN Telecoms-MailScanner: Found to be clean
X-ECN Telecoms-MailScanner-SpamCheck: not spam, SpamAssassin (not cached,
score=-100.001, required 6, autolearn=not spam, NO_RELAYS -0.00,
USER_IN_WHITELIST -100.00)
X-ECN Tele
X-ECN Telecoms-MailScanner-Information: Contact ECN Telecoms
X-ECN Telecoms-MailScanner: Found to be clean
X-ECN Telecoms-MailScanner-SpamCheck: not spam, SpamAssassin (not cached,
score=-100.53, required 6, autolearn=not spam, AWL -0.53,
NO_RELAYS -0.00, USER_IN_WHITELIST -100.00)
Below is the error, I am getting, while running my Java program through
Freeswitch dial plan.
2008-08-11 16:25:33 [NOTICE] switch_channel.c:1406
switch_channel_perform_mark_pre_answered() Ring-Ready sofia/internal/
[EMAIL PROTECTED]
2008-08-11 16:25:33 [NOTICE] mod_sofia.c:1116 sofia_receive_
Below is the error, I am getting, while running my Java program through
Freeswitch dial plan.
2008-08-11 16:25:33 [NOTICE] switch_channel.c:1406
switch_channel_perform_mark_pre_answered() Ring-Ready sofia/internal/
[EMAIL PROTECTED]
2008-08-11 16:25:33 [NOTICE] mod_sofia.c:1116 sofia_receive_
2008-08-11 17:40:53 [DEBUG] sofia.c:194 sofia_event_callback() event
[nua_r_options] status [501][Not Implemented] session: n/a
What have I not setup properly that this message constantly is displayed ona
loglevel 7 mode
Ilan Perez
___
Free
Hi Guys,
I am having this error on Freeswitch console. Any ideas.
*[ERR] switch_ivr_originate.c:926 switch_ivr_originate() Cannot create
outgoing channel of type [user] cause: [ORIGINATOR_CANCEL]*
I am using Twnikle Phone.
--
Best,
Adeel Ansari
http://www.linkedin.com/in/adeelansari
_
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