I have Cepstral working.
Can someone please tell me how to go about having it read RSS feeds? I can
have the dialplan direct it np. But I really dont have a clue how to point
it at an RSS. Any help would be great, ddint find anything in the wiki.
--
View this message in context:
*Hi,
I have seen the above mail. In that all of you tried to created dynamic
conference through diaplan itself using the database to insert the uuid,
caller_id_number, destination_number, etc .Can one guide me set the dynamic
conference and Schema for the dynamic conference.
I have tried the
Hello!
Has anybody faced such a problem with xml_curl?
2009-03-18 23:24:41 [INFO] mod_dialplan_xml.c:252 dialplan_hunt() Processing
1000-** in context default
2009-03-18 23:24:43 [ERR] mod_xml_curl.c:114 file_callback() Oversized file
detected [136089828 bytes]
2009-03-18 23:24:43
So the second issue is possibly known - really could do with a fix or a
workaround for this as we plan to use E1's for all incoming traffic.
Can anyone shed light on the first problem (extension rings for a
fraction of a second then hangs up) I mentioned below, or is that
possibly part of the
Michael Jerris There is currently no openzap (sangoma, etc) support
on windows, we hope this will be coming soon.
I found an alternative: The Linksys 3102 VoIP gateway. It's cheaper too.
Would you say the Windows port of Freeswitch is ready to be used
commercially, or I should go for a Linux
Hi all,
I'm still undecided yet whether I need proxy-media or not. As I
understand it, the only downside of enabling proxy-media is that early-
media is not possible, correct ? (Or are there other reasons why I
shouldn't use proxy-media ?)
When I disable proxy-media I get little hickups in
Pablo Hernan Saro wrote:
Hi list,
Any experience building FS in Solaris using Sun Studio?
http://www.voiceworks.pl/cypromis/tag/opensolaris/
Chris
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HarryK wrote:
I have Cepstral working.
Can someone please tell me how to go about having it read RSS feeds? I can
have the dialplan direct it np. But I really dont have a clue how to point
it at an RSS. Any help would be great, ddint find anything in the wiki.
have you tried mod_rss?
Am Thursday 19 March 2009 schrieb freeswi...@gnarg.org:
Pablo Hernan Saro wrote:
Hi list,
Any experience building FS in Solaris using Sun Studio?
http://www.voiceworks.pl/cypromis/tag/opensolaris/
Chris
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Any reason you're feeding it a 130+ meg file over XML_CURL?
/b
On Mar 19, 2009, at 6:05 AM, Леша... wrote:
Hello!
Has anybody faced such a problem with xml_curl?
2009-03-18 23:24:41 [INFO] mod_dialplan_xml.c:252 dialplan_hunt()
Processing 1000-** in context default
2009-03-18
You shouldn't use it. It has a special use case and I suspect yours
isn't it. Are you doing anything with T.38 right now?
/b
On Mar 19, 2009, at 6:57 AM, Leon de Rooij wrote:
I'm still undecided yet whether I need proxy-media or not. As I
understand it, the only downside of enabling
I'm not using T38 yet, it may be nice in the future, as long faxes
over alaw just don't work properly..
And also, there are these hickups now, that I don't have with proxy-
media enabled..
On Mar 19, 2009, at 2:15 PM, Brian West wrote:
You shouldn't use it. It has a special use case and I
Well, I guess that is something I can deal with... Actually it is for
benchmarking purposes. I was discussing about performance with a
colleague, who is a Sr Solaris Engineer, and he recommended me to
build FS in Solaris and benchmark it. He ensures that it would be
really better due to Fire
If you do a proper side by side test, let me know the results and we
will publish them.
cheers
Michal
Pablo Hernan Saro schrieb:
Well, I guess that is something I can deal with... Actually it is for
benchmarking purposes. I was discussing about performance with a
colleague, who is a Sr
Michael Jerris schrieb:
There is currently no openzap (sangoma, etc) support on windows, we
hope this will be coming soon.
Mike
On Mar 17, 2009, at 5:20 AM, Gilles wrote:
Hello
For single-host settings, getting customers to buy a separate server
just to run Freeswitch is overkill,
Hi,
maybe this message can considered off-topic, but i think can be interessing
for FreeSWITCH community.
There is a new forum on FreeSWITCH for italian people.
Please visit www.freeswitch-it.org
Any suggest are welcome
I hope to do my english a little bit better :)
Best Regards
- Andrea -
Brian,
I put two au files here:
http://www.ldr.scarlet.nl/ua-to-fs.au
http://www.ldr.scarlet.nl/fs-to-mgw.au
It's a call from a Siemens SX762 (using ALAW) to FS (no transcoding)
which bridges it to a mediagateway.
Proxy-media is disabled on the incoming sip_profile.
Both au files are
I would have to have the raw pcap to make any sense out of it.
/b
On Mar 19, 2009, at 10:34 AM, Leon de Rooij wrote:
Brian,
I put two au files here:
http://www.ldr.scarlet.nl/ua-to-fs.au
http://www.ldr.scarlet.nl/fs-to-mgw.au
It's a call from a Siemens SX762 (using ALAW) to FS (no
tone_detect! sounds good.
BTW, was there any errors in those extensions I posted. I modified something
you posted MC.
Not at first glance. What did you change?
-MC
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On Thu, Mar 19, 2009 at 4:54 AM, Gilles codecompl...@free.fr wrote:
Michael Jerris There is currently no openzap (sangoma, etc) support
on windows, we hope this will be coming soon.
I found an alternative: The Linksys 3102 VoIP gateway. It's cheaper too.
Would you say the Windows port of
On Mar 19, 2009, at 7:54 AM, Gilles wrote:
Michael Jerris There is currently no openzap (sangoma, etc) support
on windows, we hope this will be coming soon.
I found an alternative: The Linksys 3102 VoIP gateway. It's cheaper
too.
Would you say the Windows port of Freeswitch is ready
Thanks, found an install guide on the FS Wiki for libpri - will get the
server cloned then install and test.
Shall report back.
-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
Michael Collins
Sent:
I put this after the vmd tag
action application=playback data=SIT/IO_SIT.wav/
to check vmd with tones found on this page
http://en.wikipedia.org/wiki/Special_information_tone
I converted them over with Audacity to wav files and vmd worked in finding a
beep but the format was wrong for FS.
Hi,
I have some troubles with provider configuration. The are warnings in logs:
2009-03-19 19:02:48 [WARNING] mod_sofia.c:739 sofia_read_frame() We were
told to use ptime 20 but what they meant to say was 40
This issue has so far been identified to happen on the following broken
Try:
param name=inbound-codec-negotiation value=scrooge/
/b
On Mar 19, 2009, at 1:22 PM, Łukasz Czerpak wrote:
Is there any solution of this problem?
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Brian West wrote:
Try:
param name=inbound-codec-negotiation value=scrooge/
*
Unfortunately there is no difference when it is set to 'scrooge' or
other value :(
regards,
Lukasz
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Hi,
This is a known issue with some of these platforms but for
completeness can you send the actual SDP?
2009/3/19 Łukasz Czerpak luk...@czerpak.eu:
Hi,
I have some troubles with provider configuration. The are warnings in logs:
2009-03-19 19:02:48 [WARNING] mod_sofia.c:739
Brian West wrote:
what rev are you on?
trunk - ~2009-03-15 21:00
regards,
Łukasz
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The issue I seen was they invite to you with NO ptime which indicates
20ms, they should invite with ptime:60 if they want 60.
/b
On Mar 19, 2009, at 3:06 PM, Łukasz Czerpak wrote:
I've just tested g...@60i and everything works perfect - thank you
very
much. I didn't test ulaw.
What is
Thats the thing!!
Im using tcpdump to watch for packets - and i dont see any mistakes =\
The xml i sent is allright, its like a piece from my static worked xml dialplan.
But I cant understand why does FS recognise it as a 130+ mb file :D
Maybe i need to update s0mthing?)
Brian West пишет:
Any
can you do a raw request with wget?
/b
On Mar 19, 2009, at 3:48 PM, Леша... wrote:
Thats the thing!!
Im using tcpdump to watch for packets - and i dont see any mistakes =\
The xml i sent is allright, its like a piece from my static worked
xml dialplan.
But I cant understand why does FS
Brian West wrote:
The issue I seen was they invite to you with NO ptime which indicates
20ms, they should invite with ptime:60 if they want 60.
I see but there is any solution to bypass this provider's
incompatibility? I want to stay with this provider anyway - he has
very good quality
Well you said you were using G.729 for testing... when you're clearly
not... but I told you already how to fix it... for that IP or peer
g...@60i
/b
On Mar 19, 2009, at 4:03 PM, Łukasz Czerpak wrote:
I see but there is any solution to bypass this provider's
incompatibility? I want to stay
Its not the easy thing. But what I can do is to attach here full tcpdump log,
with all packets.
Brian West пишет:
can you do a raw request with wget?
/b
On Mar 19, 2009, at 3:48 PM, Леша... wrote:
Thats the thing!!
Im using tcpdump to watch for packets - and i dont see any mistakes =\
Are you setting a Content Length header in the HTTP response??
2009/3/19 Леша... qu...@mail.ru:
Thats the thing!!
Im using tcpdump to watch for packets - and i dont see any mistakes =\
The xml i sent is allright, its like a piece from my static worked xml
dialplan.
But I cant understand
As I see theres only :
Content-Type: text/html; charset=utf-8
But no Content Length
SP пишет:
Are you setting a Content Length header in the HTTP response??
2009/3/19 Леша... qu...@mail.ru:
Thats the thing!!
Im using tcpdump to watch for packets - and i dont see any mistakes =\
The xml i
(sorry for the broken thread: I don't know how to avoid this when
answering through the digest version of the mailing list)
Michael Jerris You could use Netborder Express with it.
Thanks for the tip. I didn't know this device. I'm not sure I
understand the difference between this PCI card and
I'm doing what you want to do and using SPA3102.
It's much easier to get someone to try it this way when dealing with small mom
and pop size business.
Haven't tried higher concurrent call volumes with some of the PCI cards
mentioned.
If you haven't done this already, my advice is first to
I guess that's why they call us noobs! heh ;)
Working perfectly, thank you!!
Raymond Chandler-2 wrote:
HarryK wrote:
I have Cepstral working.
Can someone please tell me how to go about having it read RSS feeds? I
can
have the dialplan direct it np. But I really dont have a clue
Ok I got Skypiax working just fine but there is no audio either way when I
say call into a conf using the Skype username.
I had this no audio problem with NAT when I first setup FreeSWITCH and
solved it by using Scenario 2 from this wiki page...
I don't think no sound is caused by NAT, better to check sound driver
and configuration.
On Mar 20, 2009, at 10:53 AM, HarryK wrote:
Ok I got Skypiax working just fine but there is no audio either way
when I
say call into a conf using the Skype username.
I had this no audio problem
Hi Anthony,
I installed the patch, but I don't think it accomplishes what I want.
I want the opposite, I want the fifo caller to continue along with the
dialplan after the agent (consumer) is finished with servicing the call.
This might be useful in situations where there could be an IVR
I wrote this wiki page a while back. Did it help?
http://wiki.freeswitch.org/wiki/Mod_rss
On Thu, Mar 19, 2009 at 2:41 AM, HarryK switchser...@gmail.com wrote:
I have Cepstral working.
Can someone please tell me how to go about having it read RSS feeds? I can
have the dialplan direct it
http://jira.freeswitch.org/browse/MODASRTTS-11
Might wanna know about that issue also :)
/b
On Mar 20, 2009, at 12:02 AM, Carlos Talbot wrote:
I wrote this wiki page a while back. Did it help?
http://wiki.freeswitch.org/wiki/Mod_rss
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