Hi Michael
Is the ztcfg output supposed to say something like 2 channels configured?
I have not set up openzap yet because I don't know if the openzap.config is
good or not.
Can you give me any suggestion if the ztcfg output I am getting is proper or
not?
thx,
woody
-MC
>
> On Wed, Oct 7, 200
Hi,
sofia_reg_handle_register, once it got executed, actuallyl where it will
maintain the registred users adn groups data,
Thanks
Srinivasula Reddy K
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Brian West wrote:
> From what I have been told h323plus is a based/fork of OpenH323 which
> OPAL is just a continuation of OpenH323.
>From a quick search of gmane.org, the situation seems a little more
complicated.
http://article.gmane.org/gmane.comp.telephony.openh323.general/10930/match=h
Hi,
currently I'm playing around with call pickup and Snom phones.
I'm using the intercept function for that.
My "problem" is now that after the call pickup (which works fine) I don't see
the caller id of the original call.
Instead I see the pickup code, e.g. *820
I've tried to change nearly ev
> Can you advise sip-providers offering t38?
Gafachi has T38 fax support.
On Oct 7, 2009, at 4:26 PM, Vladimir Elizarov wrote:
> Kristian Kielhofner пишет:
>> Since no one else has responded I'll chime in with some general
>> advice.
>>
>> It's troubling to see that your "provider" is using A
Kristian Kielhofner пишет:
> Since no one else has responded I'll chime in with some general advice.
>
> It's troubling to see that your "provider" is using Asterisk to face
> you (the customer).
>
> I've never had any luck getting T.38 to work (at all, in any mode)
> using Asterisk. I've heard of
From what I have been told h323plus is a based/fork of OpenH323 which
OPAL is just a continuation of OpenH323. So why not support the
developers of OPAL/OpenH323 ?
/b
On Oct 7, 2009, at 7:50 AM, Georgiewskiy Yuriy wrote:
> We are developing module to handle h323 proto now, we try to use
Since no one else has responded I'll chime in with some general advice.
It's troubling to see that your "provider" is using Asterisk to face
you (the customer).
I've never had any luck getting T.38 to work (at all, in any mode)
using Asterisk. I've heard of other people making it work but
consid
Double check your firewall and:
1) ensure you've actually enabled UPNP
and
2) Ensure that any mention of a SIP ALG (application level gateway) is
turned off. SIP ALGs tend to really screw things up.
On Wed, Oct 7, 2009 at 12:48 PM, Andy wrote:
> I've now set the external ips manually to be my e
Thanks Brian,
I've now set the external ips manually to be my external ip and have forward
all ports through my firewall to the FS server. It's actually set up as a
DMZ to everything is being forwarded without restriction but sadly DTMF and
HANGUP messages are still not getting through. Have I mi
s/auto-nat/$realip/ then forward the rtp ports and sip ports.
/b
PS chances are you have to ENABLE upnp.
On Oct 7, 2009, at 12:58 PM, Andy wrote:
Many thanks Brian, the firewall docs assure me it is uPnp but is
probably lying or a poor implementation. Could you point me to the
right section
Many thanks Brian, the firewall docs assure me it is uPnp but is probably
lying or a poor implementation. Could you point me to the right section of
the Wiki to tell me how to do this manually as I've been scouting for some
time and can;t seem to find the right thing. sorry if I'm being blind.
_
2009/10/7 srinivasula reddy :
>
> Hi
>
> can any please tell me where registered calls are stored, so when incoming
> call came to mod_sofia.c how it will check it is registered or not?\\
I use 2 profile
internal profile require register before make a call.
external i use odbcquery check ipa
No its because you're not behind a upnp/nat-pmp router so you'll have
to manually forward everything... All the info you showed displaying
the profile status is correct.
/b
On Oct 7, 2009, at 12:44 PM, Andy wrote:
Is this because I'm using port 5060 externally?
Cheers
Andy
_
Thanks Brian, sorry should have pre-empted that one as I've issue before.
start_dtmf is in the dialplan and occurs at the start of every call.
On closer inspection however it appears that only part of the nat setup is
taking place.
sofia status gives:
Name Type
Hello,
I read in the wiki that binding blocks are processed in sequential order
in a failover matter.
So I created the following bindings for XML-Curl:
However grepping the network traffic I can see that Freewitch always
fetches both servers fo one binding. So there is no real failover.
How can
The ztcfg seems okay since it set the signaling to kewlstart. What part is
not working? Be sure to pastebin your openzap.conf and openzap.conf.xml
files as well as a full debug of what happens when loading mod_openzap.
-MC
On Wed, Oct 7, 2009 at 3:40 AM, Woody Dickson wrote:
> Hi,
>
> I am trying
On Wed, Oct 7, 2009 at 2:40 PM, Claudiu Filip wrote:
>
>
>
> Hi Tihomir,
>
>
> I've done some tests to see how suitable is freeswitch as a
> SIP/H323 translator and you are right about the fact that H323
> 'alert+open logical channel' will generate a SIP '200 OK'. I was
>
On Wed, Oct 7, 2009 at 3:34 AM, lakshmanan ganapathy
wrote:
> Hi,
> Again I was struck in a problem, Here is the scenario.
>
> On incomming call, I just call an event outboud socket. But what happens
> is, for the first 15 call, it is working fine. But from the 16th call to
> 30th call, it says th
Claudiu Filip wrote:
> Hi,
>
> Wednesday, October 7, 2009, 3:58:20 AM, Anthony M. wrote:
>> barely get anybody asking about h.323.
>
>
> H323 may not be popular for small ITSPs or small/medium PBXes, but
> it's widely used by the big players.. and freeswitch doesnt share the
> same goals with a
Hi
can any please tell me where registered calls are stored, so when incoming
call came to mod_sofia.c how it will check it is registered or not?\\
--
Srinivasula Reddy K
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You are missing the point, it is only rejecting auth for FS, Asterisk and
X-Lite work fine with the same config for that gateway.
On Wed, Oct 7, 2009 at 10:20 AM, Brian West wrote:
> I would suspect its a PEBKAC. I mean if you could register to a
> gateway that rejected auth... what purpose wo
You'll need to set presence_id so it can work properly. SEE the
default config that does exactly that with sofia_contact in the dial-
string on the domain.
/b
On Oct 7, 2009, at 8:26 AM, Michael Jerris wrote:
When calling the Bridge application with data parameter of sofia/
profile-name/num
Bet you its inband dtmf and you need to start the dtmf detector.
/b
On Oct 7, 2009, at 8:11 AM, Andy wrote:
I can hear the IVR message played down the phone line so outgoing
audio is ok.
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No!
/b
On Oct 7, 2009, at 8:37 AM, Muhammad Shahzad wrote:
> Great. Just added it. Is there any user limit on this?
>
> Thank you.
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I would suspect its a PEBKAC. I mean if you could register to a
gateway that rejected auth... what purpose would auth serve in the
first place?
/b
On Oct 7, 2009, at 8:48 AM, Nicolas Brenner wrote:
> Is there some way to make FS register with the gateway that is
> rejecting the authentica
I think the way to determine the funding is to get all the most important
issues up on jira, try to deal with them and see if we need to put bounties
on any of them to get them done faster.
On Wed, Oct 7, 2009 at 3:37 AM, Tihomir Culjaga wrote:
> Anthony,
>
> of course, nobody wants to start an
I am not commenting on how popular it is overall.
I am commenting on the specific demand presented to us.
I don't know what else to say to explain that I am completely neutral when
it comes to this topic.
One more time:
1) We are an open source project who volunteer most of our time as well as
pa
Is there some way to make FS register with the gateway that is rejecting the
authentication? is it FS or the SIP server at fault? Why would X-Lite work
and FS not?
Thanks again for your time and help.
On Tue, Oct 6, 2009 at 5:46 PM, Brian West wrote:
> btw My mistake it doesn't assume auth it
Great. Just added it. Is there any user limit on this?
Thank you.
On Wed, Oct 7, 2009 at 6:15 PM, Chris Chen wrote:
> Hi Muhammad, the simple and reliable solution for you where SIP is being
> blocked is add
> conf+...@conference.freeswitch.orgto
> your Goolgetalk buddy list, and you can call
On Oct 6, 2009, at 4:14 AM, Yehavi Bourvine wrote:
Hello,
We have Polycom and SNOM phones running with FreeSwitch. The
Polycoms have shared lines defined and the SNOMs have both shared
lines and BLFs (defined as extensions in the phone config). I've
tried supporting both, but have some
Incorrect NAT configuration so one of the boxes is not actually
getting a BYE.
On Oct 5, 2009, at 3:13 AM, Maciej Aniserowicz wrote:
Hi,
When I use two FreeSWITCH instances ('internal' and 'external'), all
users register to the 'external' instance which acts as a gateway by
'internal' in
switch_ivr_async.c:480
On Oct 5, 2009, at 3:16 AM, Maciej Aniserowicz wrote:
Hi,
When I record a call in FS, it only creates a 388-byte-long wav
file. The conversation is no written there, and FS deletes the file
when the session finishes.
What can cause this strange behavior?
__
Hi Muhammad, the simple and reliable solution for you where SIP is being
blocked is add
conf+...@conference.freeswitch.orgto
your Goolgetalk buddy list, and you can call from there to join the
conference, simple and straightforward.
Chris
On Wed, Oct 7, 2009 at 12:24 AM, Muhammad Shahzad <
shahe
Hi,
Wednesday, October 7, 2009, 3:58:20 AM, Anthony M. wrote:
> barely get anybody asking about h.323.
H323 may not be popular for small ITSPs or small/medium PBXes, but
it's widely used by the big players.. and freeswitch doesnt share the
same goals with asterisk.
Best wishes,
Claudiu Fil
Hi folks,
I know this problem comes up all the time so sorry to bring it up again but
I can't seem to find the answer in previous posts.
I have my freeswitch installation behind a DLink firewall so the freeswitch
server is natted. I have added what I believe are all the necessary rules to
the f
On 2009-10-07 01:41 +0200, Tihomir Culjaga wrote freeswitch-us...@lists.fre...:
We are developing module to handle h323 proto now, we try to use mod_opal and
try improve it, but no luck,
there is many issues in libopal, and finaly we now move to h323plus library.
TC>Diego,
TC>
TC>what i'm point
What codecs are all the call legs using, also, please try current svn
trunk.
Mike
On Oct 7, 2009, at 3:39 AM, Maciej Aniserowicz wrote:
>
> Sorry about posting several questions at once, I wasn't aware it's
> "rude".
> Let's concentrate on this issue then.
>
> I use FS rev 14994. Phones on e
Hi Tihomir,
I've done some tests to see how suitable is freeswitch as a
SIP/H323 translator and you are right about the fact that H323
'alert+open logical channel' will generate a SIP '200 OK'. I was
able to fix that with a couple of changes in mod_opal.cpp, howeve
Sorry about posting several questions at once, I wasn't aware it's "rude".
Let's concentrate on this issue then.
I use FS rev 14994. Phones on extensions:
1) x-lite
2) cisco sip phone
3) audio played by fs to the extension being eavesdropped
I did not change any codec configuration, I just
Hi,
Thanks for replying. The advise u gave resolved my problem. I want to ask
three questions related to nibble bills, as I 'm listing down below;
1- Can we select/use dynamic tables for billing using nibble bill?
2- Can we define more than two tables and attributes in nibblebill.conf.xml?
3- As
[debian :~]# ztcfg -vvv
Zaptel Version: 1.4.11
Echo Canceller: MG2
Configuration
==
SPAN 1: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1)
Channel map:
Channel 01: Clear channel (Default) (Slaves: 01)
Channel 02: Clear channel (Default) (Slaves: 02)
Channel 03: Clear chann
lakshmanan ganapathy said:
> On incomming call, I just call an event outboud socket. But what
happens is,
> for the first 15 call, it is working fine. But from the 16th call to 30th
> call, it says the below error.
What is displayed with
ztcfg -vv
What is displayed with
cat /proc/zaptel/1
Hi,
I am trying to setup a Digium TDM400P following the instruction on the
wiki.
It is a 1 fxo and 1 fxs card, so I tried
loadzone=in
defaultzone=in
fxsks=2
fxoks=1
and
loadzone=in
defaultzone=in
fxsks=1
fxoks=2
None works. Does anyone know how it should be configured?
Here is what I get by f
Hi,
Again I was struck in a problem, Here is the scenario.
On incomming call, I just call an event outboud socket. But what happens is,
for the first 15 call, it is working fine. But from the 16th call to 30th
call, it says the below error.
2009-10-07 15:07:48.201846 [WARNING] ozmod_libpri.c:761
Here are the details
openzap.conf
[span zt PRI_1]
trunk_type => e1
b-channel => 1:1-15
d-channel=> 1:16
b-channel => 1:17-31
Zaptel.conf
span=1,1,0,ccs,hdb3
bchan=1-15,17-31
dchan=16
loadzone= us
defaultzone = us
mercutioviz wrote:
>
> Pastebin your openzap.conf file. Also, is
Hi List,
maybe someone can give me some hints to get faster into this stuff. Here
the use case:
I am Ext A and talking to Ext B. Now Ext C is calling to another number
: D.
How can I
1. Pic up the calling Ext C ?
2. Include it to my current call to Ext A ?
I would need some infos wich elem
Anthony,
of course, nobody wants to start anything... we are all here to help making
FS a better product.
so, regarding the founding for mod_opal ... what is the amount you need?
Tihomir.
On Wed, Oct 7, 2009 at 2:58 AM, Anthony Minessale <
anthony.miness...@gmail.com> wrote:
> I didn't mean
On Wed, Oct 7, 2009 at 9:53 AM, Moiz Chinoy wrote:
> Thanks for your replies
>
> gsmopen,org seems interesting but it does not have any documentation.
> Can anyone point me where I can find information regarding this
> project.
>
:)
it is prealpha now, will available as alpha in couple week o
Thanks for your replies
gsmopen,org seems interesting but it does not have any documentation.
Can anyone point me where I can find information regarding this
project.
On Tue, Oct 6, 2009 at 4:04 PM, Seven Du wrote:
> maybe you can check this: http://www.gsmopen.org/
>
> 2009/10/6 Moiz Chino
srinivasula reddy wrote:
>
> Thank you very much for your reply, i have gone through the link what u
> have sent me, they have given how to access remote system through webserver,
> but they have not given how to access from local system,
On that page, there is an example of how to use mod_curl
Hi brain west,
Thank you very much for your reply, i have gone through the link what u
have sent me, they have given how to access remote system through webserver,
but they have not given how to access from local system,
if u know please hellp me
Thanks
On Wed, Oct 7, 2009 at 12:22 PM, Brian W
http://wiki.freeswitch.org/wiki/Mod_xml_curl Is a great start.
/b
On Oct 7, 2009, at 1:43 AM, srinivasula reddy wrote:
Hi,
i want use mod_xml_curl, the xml files also there in my local
system, i dont want to take from any other system,
can any please tell me how to configure mox_xml.conf.x
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