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
___
Freeswitc
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 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 do
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 recording
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 probl
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...
http://wiki.freeswitch.org/wiki/General_NAT_exam
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 hav
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 s
(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
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... :
>> Thats the thing!!
>> Im using tcpdump to watch for packets - and i dont see any mistakes =\
>> The xml i sent
Are you setting a Content Length header in the HTTP response??
2009/3/19 Леша... :
> 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 F
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 mista
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 s
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 qualit
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 doe
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 r
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
Brian West wrote:
> Well you can't have ptime 60 one way and 20 the other it just won't
> work. Also I can't even think that this illegal codec was even tested
> at 60ms... Try it with ulaw and see what it does. or only allow
> g...@60i and see what it does.
>
I've just tested g...@60i and e
I have phones registered to a FS box, and an * box. There is a sip trunk
between the two boxes.
A phone on my * (54321) calls a FS phone (12345); if I hang up the * phone
while it's still ringing, this is what I get on the sip trace on FS:
...
2009-03-19 15:05:40 [NOTICE] switch_ivr_originate.c
Brian West wrote:
> what rev are you on?
>
trunk - ~2009-03-15 21:00
regards,
Łukasz
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Well you can't have ptime 60 one way and 20 the other it just won't
work. Also I can't even think that this illegal codec was even tested
at 60ms... Try it with ulaw and see what it does. or only allow
g...@60i and see what it does.
/b
On Mar 19, 2009, at 2:20 PM, Łukasz Czerpak wrote:
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 :
> 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
what rev are you on?
/b
On Mar 19, 2009, at 1:52 PM, Łukasz Czerpak wrote:
>> *
>
> Unfortunately there is no difference when it is set to 'scrooge' or
> other value :(
>
>
> regards,
> Lukasz
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Brian West wrote:
> Try:
>
>
> *
Unfortunately there is no difference when it is set to 'scrooge' or
other value :(
regards,
Lukasz
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Try:
/b
On Mar 19, 2009, at 1:22 PM, Łukasz Czerpak wrote:
Is there any solution of this problem?
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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
platforms/dev
I put this after the "vmd" tag
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.
However, after I switch the format of the
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:
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
On Thu, Mar 19, 2009 at 4:54 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 r
On Thu, Mar 19, 2009 at 2:08 AM, Mark Tabron
wrote:
> 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 han
> 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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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 transcodi
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 extracted
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 -
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 Freeswit
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 Sol
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 Engine
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
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 prox
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
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
>
>
> ___
> Freeswitch
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_
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 t
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
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 [ERR
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 same
*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 abov
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.
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