In the previous reply you told me to use new OUTBOUND connection.
But in this post you mention INBOUND connection.
That confusion only made me to ask the question once again. Pardon me if I
made any mistake.
Making a new inbound connection does the task. Thanks for that.
On Sat, Nov 28, 2009
Hi All,
I am currently installing a Sangoma A102 card to work with FS using wanpipe
drivers (OS = Ubuntu Jaunty). The problem is I can't get openzap-related
modules to compile:
cd wanpipe-3.5.6.5/
make openzap
...
make[2]: Leaving directory `/usr/src/wanpipe-3.5.6.5/api/libsangoma'
make[1]:
I did manage to build these drivers, but maybe you're not doing it the
right way. Sangoma document state that the drivers should be built by using
their ./Setup script that does all that is required.
I did use ./Setup
install which builds the kernel modules, the wanrouter utilities and
install
make openzap is the correct way to build when using with openzap/freeswitch.
If you are having issues with this you should check with sangoma support as to
why that build of the drivers is not supporting it properly and what version
you should be using.
Mike
On Nov 30, 2009, at 5:41 AM,
Hi All,
Thought I would share my solution to this DTMF problem: it turns out my
ISP was capping my bandwidth dropping packets to keep the connection
1Mbps, so the experienced DTMF loss was actually packets being discarded.
On my way to this discovery I tested Freeswitch DTMF quite
Hi,
Is there anyway to detect when a channel is park in a way that is similar to
hangup-hook or answer-hook? I would like to detect that inside a custom
mod, without using the event mechanism?
woody
___
FreeSWITCH-users mailing list
Hello,
I'm using Freeswitch with a Sangoma A400 card, and I'm having
CLIP problems on the FXS ports.
When I ring on FXS ports, the connected
phone does not display callerid/callerid-name.
I tried turning the stuff
of in openzap.conf.xml () but it did not help.
As a side note, turning
On Mon, Nov 30, 2009 at 4:49 AM, Neil Patel ne...@cs.stanford.edu wrote:
Hi All,
I am currently installing a Sangoma A102 card to work with FS using wanpipe
drivers (OS = Ubuntu Jaunty). The problem is I can't get openzap-related
modules to compile:
cd wanpipe-3.5.6.5/
make openzap
...
can you test svn trunk or latest pre release of 1.0.5
On Mon, Nov 30, 2009 at 9:36 AM, François Legal de...@thom.fr.eu.orgwrote:
Hello,
I'm using Freeswitch with a Sangoma A400 card, and I'm having CLIP problems
on the FXS ports.
When I ring on FXS ports, the connected phone does not
Hello,
has anyone of you tried the RTP TAP function of sangoma`s wanpipe driver?
It is described here: http://wiki.sangoma.com/wanpipe-voice-rtp-tap
On my side wanrouter log says that RTP TAB is configured and enabled,
but I can't detect any udp packets received by the remote server (which
The correct way to pass non-standard headers is X- not X_ .
action application=set data=accountcodec=${sip_h_X-ACCOUNTCODE} /
On Sat, Nov 28, 2009 at 12:47 PM, Simon Woodhead simon.woodh...@me.com wrote:
Hi folks,
I'm hoping someone can help me get at custom headers in the dial-plan. I've
I'm attempting to configure several varieties of polycom (SoundPoint IP
550, SoundPoint IP 601) phones to connect to a freeswitch instance using
a domain other than default (i.e. the ip address). Everything works
wonderfully as long as the domain is named exactly the same thing as the
server
I don't quite understand what you are talking about?
So you have bypass_media=true and you attempt to make an attended xfer
as soon as you complete the transfer according to your trace FS does
re-invites to convert the call to be exchanging media with FS. The o= lines
you don't like are being set
Hello Helmut,
On Mon, Nov 30, 2009 at 11:52 AM, Helmut Kuper helmut.ku...@ewetel.dewrote:
Each try to do some kind of printf debugging in wanpipe-driver doesn't
succeed.
Any ideas?
The way the rtp tapping works right now is kinda hackish and pretty much
Asterisk/Zaptel-based. We depend on
Hi to all,
shortly i'll make a pbx for a customer that uses a couple of isdn bri
lines and, looking for hardware, i've seen that not too much expensive
isdn cards that works well are the ones that uses hfc-4/8s controller
(specifically i'll use a OpenVox B200P that has 2 ISDN ports and use an
Tony committed my patch for doing 'week of month' conditions in the XML
dialplan along with some holiday routing examples to the default
dialplan. Now you can detect all the major US holidays in pure dialplan
XML without having to do any nasty math or anything (I did it all for
you).
I've also
Thanks for this goodness. I am sure to use it so it is appreciated.
On Mon, Nov 30, 2009 at 2:51 PM, Andrew Thompson and...@hijacked.us wrote:
Tony committed my patch for doing 'week of month' conditions in the XML
dialplan along with some holiday routing examples to the default
dialplan. Now
Hello,
I would like Freeswitch to pass the Remote-Party-ID field of the called
party (sent in the Ringing OK when answering the call) back to the
originator's phone. How can I do that?
The drive for this is: Our Freeswitch is connected via a Cisco gateway and
PRI to the university's phone
Just set the variables effective_callee_id_name and
effective_callee_id_number in your dp before you answer the call
On Dec 1, 2009 12:08 AM, Yehavi Bourvine yehavi.bourv...@gmail.com
wrote:
Hello,
I would like Freeswitch to pass the Remote-Party-ID field of the called
party (sent in the
Hello Anthony,
I think I did not explain myself correctly: The destination sends the
Remote-Party-ID in the Ringing and OK replies, but they are not relayed to
the original caller.
Thanks! __Yehavi:
2009/12/1 Anthony Minessale anthony.miness...@gmail.com
Just set the
Are you on SVN trunk? As far as I recall the callee_id_number/name
stuff isnt in 1.0.4.
Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mr...@avgs.ca
On 1-Dec-09, at 1:42 AM, Yehavi Bourvine wrote:
Hello Anthony,
I think I did not
Are you on SVN trunk? As far as I recall the callee_id_number/name stuff
isnt in 1.0.4.
No, because the SVN has problems with Emailing the voicemail...
We use 1.0.4 and set sip_callee_id_number/name which works. I would like to
not set it and get it from the other side...
22 matches
Mail list logo
