Hello Ognjen,
From the tests I've done it is not so... When I set the profile to use
INFO, and a phone calls and asks for RFC2833 (phone-events in the SDP) the
FreeSwich ignores it (does not have phone-events field in the reply SDP)
which causes the phone to not send RFC2833 events...
Hi Adam
Excellent first steps!
Thankyou for the hint.
Now I hope somebody can tell me what I'm doing wrong next...
I've gotten it to register to the testprovider here (musimi.dk), but I get an
error when I create an account for testing with the X-Lite phone.
It displays 403 forbidden in the
You previously stated that your Cisco gateway has some "bug" that
prevents you from using RFC2833, did you enable "dtmf-relay rtp-nte" on
the voip dial-peer that the call is using?
Unless you have configured the Cisco to support assymetric SDP or are
using a non-default "rtp payload-type nte" s
Hello Metik,
2009/12/6 Metik
> You previously stated that your Cisco gateway has some "bug" that
> prevents you from using RFC2833, did you enable "dtmf-relay rtp-nte" on
> the voip dial-peer that the call is using?
>
>
It is a PSTN dialpeer here, and it cannot be defined on it...
> Unless y
Concerning,
> Which I'm kinda confused about, I don't have any 192.168 net here??
I think, this is a default entry in the acl.conf.xml. Please check the
entries there. But normally this shouldn't stop freeswitch from working
and handling requests.
Can you set the console_log_level to "debug" in va
Dear all
Some feedback regarding using Lua to access core database:
First of all, I did not succeed to get SQLite drivers in Lua or ODBC-drivers in
Lua to work. The SQLite driver did compile OK, but there was an error when
loading into Lua. The ODBC driver did also compile OK, did load into Lua
Hello,
I would like to offer a dictation service to a secretary.
Means:
* the boss is dictating some text on a certain phone number
* the secretary picks up the recording on the phone and types the
text into the computer
As the secretary is not able to type in as fastly as heir bos
Jon,
What version of MySQL are you using?
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Hi Peter
Ok, I got the net changed in acl.conf.xml.
I then tried setting console_loglevel, but I don't see any output on the
console, it could very well be because it's a FreeBSD, and has very limited
console.
But after a restart it registers!
So for some reason it needed a nudge there, very in
Hi,
I just checked the SIP traces and it looks like FS sends a sipfrag message to
the phone with
caller_id_name and caller_id_number instead of effective_caller_id_name and
effective_caller_id_number values.
Thanks, Klaus
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch
Hi Klaus,
Try setting ignore_display_updates=false
Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mr...@avgs.ca
On 6-Dec-09, at 1:38 PM, Klaus Hochlehnert wrote:
Hi,
I just checked the SIP traces and it looks like FS sends a sipfrag
m
Unless the IOS you are running is extremely buggy, "debug voip ccapi"
commands should not provide you with that detail, what you really want
to use is "debug voip rtp session named-event".
Normal SIP-to-PSTN calls should use both a pots and voip dial peer but
DTMF relay type is determined by t
Someone else was asking about this too.
I could probably write a dictaction mod in c like the one I made for
asterisk starting at about $3k depending on the featureset required.
On Dec 6, 2009 10:30 AM, "Peter P GMX" wrote:
Hello,
I would like to offer a dictation service to a secretary.
Means:
Or set it to true depending on the case
Also consider using set_profile_var to set the caller id explicitly instead
of using effective. There is also effective_callee_id name and number you
could set on the a leg. You'll have to expirement but the one mathieu said
is your best bet.
On Dec 6, 200
Some more bad news for you, info dtmf spec has expired and has been
abandoned. Wait till you see what they did accept instead..
On Dec 6, 2009 1:22 PM, "Metik" wrote:
Unless the IOS you are running is extremely buggy, "debug voip ccapi"
commands should not provide you with that detail, what
Hello,
I have a few questions about Ploycom's usage and provisioning for which I
found no answers neither at the docs nor on the WEB:
- I would like to enable SIP/TLS. for this I have to import the root
certificate. How can I do it via the XML config files? the only method I
found is
Most of this is unfortunatly because you do not have the proper skill to set
it up because, with the proper skills, all of the ways you tried would have
ended sucessfully. I say that beacause I have had many users use each of
the different methods in your list of failures only they were sucessful.
Ok, set_profile_var did the trick and also works with intercepted calls.
Thanks, Klaus
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Anthony
Minessale
Sent: Sunday, December 06, 2009 8:32 PM
To: freeswitch-users@lists.fre
>Registrations:
>=
>=
>As far as I can see, everything looks ok, except for the
>2009-12-06 19:35:43.111477
The MySQL version is 5.1.37. Well I'm not an expert on every field, and I have
no skills in the C, include libraries, and the art of compiling. For this I
have to follow the guidelines. But it wouldn't harm the FS project if it
generally became more accessible to the race of non-specialists, whi
Oh that's a lot of money,
anybody else needs this feature, so we may share a bounty?
Best
regards
Peter
Anthony Minessale schrieb:
>
> Someone else was asking about this too.
> I could probably write a dictaction mod in c like the one I made for
> asterisk starting at about $3k depending on the
Yes, exactly my point.
Like I said you have several choices be paitent till we have time to
code it for free, post a bounty to increase the chance somone will do it
from the community, hire someone to set it up for you or keep trying
yourself.
Did I miss something?
On Dec 6, 2009 3:38 PM, "J
Hi Joseph
Ahh, yes, that got rid of that error :-)
Now on to the next one.
So now it's connecting, both at my provider, and my softphone.
Now I have to figure out why it tells me 'Call failed: not found' when I try to
call out of the system...
But I think that's a task for tomorrow when I'm mor
Greetings. We need to add database access to an IVR application we are
prototyping. Based on FS "best practice" suggestions, we are using Lua for
the scripts. Since FS uses SQLite internally, we presumed that Lua + SQLite
would be a recommended approach. However, we can't find any examples of this
Greetings. We are attempting to add sqlite access to an IVR application we
are prototyping. We are using lua for the scripts. Is there an example
anywhere of a lua + sqlite script? Do we need to install luasql? Any
help/pointers greatly appreciated.
--Steve Klein
___
On Mon, Dec 7, 2009 at 9:07 AM, Steve Klein wrote:
> Greetings. We need to add database access to an IVR application we are
> prototyping. Based on FS “best practice” suggestions, we are using Lua for
> the scripts. Since FS uses SQLite internally, we presumed that Lua + SQLite
> would be a recomm
I have 2 identical Dell PowerEdge 1950 servers running FreeSWITCH Version 1.0.4
(exported) with only one thing difference which is the first one is running
with -hp enabled; however, I have noticed that the one with -hp option consumed
double in memory usage than the other one.
I wonder wheth
This bug has been now closed out in jira due to no response for requested
information. If you wish to resolve this issue please follow up on your bugs
when information is requested.
Mike
On Oct 12, 2009, at 12:04 PM, Maciej Aniserowicz wrote:
>
> Nope, I wanted to make sure that this is inde
Guys,
im after info from people with experience with AudioCodes Mediant 2k PRI
Gateways.
specifically how well they inter-op with Freeswitch, and how compliant their
SIP stack is.
I guess the bottom line is, would you recommend these gateways or would you
suggest something else ?
--
Sincerely
Hi,
I know there's some chang on att_xfer, and after upgrade(re-bootstrap)
to trunk code, no sound after att_xfer.
Then I rebuild FS 15807 with a fresh checkout, but still using the old
conf/ settings, sound is ok, but there are other problems:
A call B, and B att_xfer C
1) origination_cancel_k
Please report bugs to jira.freeswitch.org.
Mike
On Dec 6, 2009, at 11:45 PM, Seven Du wrote:
> Hi,
>
> I know there's some chang on att_xfer, and after upgrade(re-bootstrap)
> to trunk code, no sound after att_xfer.
>
> Then I rebuild FS 15807 with a fresh checkout, but still using the old
> c
We are using Audiocodes and Sangoma netborder express GW with
Freeswitch . it works well.
Thanks
Imthiyaz
On Mon, Dec 7, 2009 at 9:34 AM, jay binks wrote:
> Guys,
> im after info from people with experience with AudioCodes Mediant 2k PRI
> Gateways.
> specifically how well they inter-op with F
Pardon me if this has been addressed already.
How does one go about having in the simplest instance 2 servers
registering with each other on startup whereby the users registering
would be able to call each other.
The 2 servers are in different domains.
Thanks.
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