Hi
I tried to download non-US sound files, but I am getting this error:
--21:50:09-- http://files.freeswitch.org/freeswitch-sounds-fr-1.0.10.tar.gz
Resolving files.freeswitch.org... 69.174.57.101
Connecting to files.freeswitch.org|69.174.57.101|:80... connected.
HTTP request sent, awaiting
On 24/08/09 4:27 PM, João Mesquita wrote:
I need to work a little bit more on that documentation as well.. I saw a
few conflicts with the core documentation too. Will get there once I
have some more time left.
Awesome man - thanks for these.
So, one more question - in the show calls (or show
On 24/08/09 6:00 PM, Juan Backson wrote:
Hi
I tried to download non-US sound files, but I am getting this error:
--21:50:09-- http://files.freeswitch.org/freeswitch-sounds-fr-1.0.10.tar.gz
Resolving files.freeswitch.org... 69.174.57.101
Connecting to files.freeswitch.org
Hi Vladimir,
Did you get an update on this at all?
Thanks
vmorales wrote:
Hi Michal,
Just checking in to see if you've been able to take a stab at this.
Thanks,
Vladimir
-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
On Mon, Aug 24, 2009 at 5:31 AM, Brian Westbr...@freeswitch.org wrote:
It requires internet connectivity. It calls a remote system to play
which is out of our control.
Yeah, I noted this too, since a couple weeks at least...
Maybe let's Todd know it's monkeys are out of voice?
-giovanni
Hello,
is there any chance to limit the listening ips of the xml-rpc server
(which is currently 0.0.0.0) to another one (e.g. 127.0.0.1)?
Best regards
Peter
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Me too :)
- Original Message -
From: Brian West br...@freeswitch.org
To: freeswitch-users@lists.freeswitch.org
Sent: Monday, August 24, 2009 1:53 AM
Subject: Re: [Freeswitch-users] zrtp endpoints have different sas through
fs1.0.4
Wish they would send me one for my E63 for testing...
No french files exist yet. ;)
/b
On Aug 24, 2009, at 1:00 AM, Juan Backson wrote:
Any idea what's wrong?
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I updated testclient.c so you can see how now.
#include stdio.h
#include stdlib.h
#include esl.h
int main(void)
{
esl_handle_t handle = {{0}};
esl_connect(handle, localhost, 8021, ClueCon);
esl_send_recv(handle, api status\n\n);
if (handle.last_sr_event
try looking for the channel_originate events which is fired as soon as any
outbound call is mature enough to have variables.
On Mon, Aug 24, 2009 at 12:57 AM, Mikhail Krivushin
m.krivus...@imarto.netwrote:
I put some variable in originate command for identify created channel for
my app using
Hi,
Does anyone know the purpose of fifo_orbit_announce?
When does fifo_orbit_announce get played?
Thanks,
JB
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You're using Proxy Media and the only clean way to do this is not use
proxy media that way a complete clean SDP will be generated for the B-
Leg.
/b
On Aug 24, 2009, at 9:45 AM, Hound Dog wrote:
is there a way to have FS build a clean SDP message ?
carriers need topology hiding , its an important feature for both security and
also to hide you business partners from each other
freeSwitch talks about it and also does a good job in hiding the signalling
topology
there is however a hole in the SDP manipulation that I am trying to plug and
But that would allow tell FS to also do transcoding in some cases, correct?
Is there a way to avoid transcoding and still build the required SDP in
a clean manner?
Best regards,
Vlasis Hatzistavrou.
Brian West wrote:
You're using Proxy Media and the only clean way to do this is not use
You can disable transcoding on the sofia profiles... see defaults.
/b
On Aug 24, 2009, at 10:28 AM, Vlasis Hatzistavrou (KTI) wrote:
But that would allow tell FS to also do transcoding in some cases,
correct?
Is there a way to avoid transcoding and still build the required SDP
in
a
if you establish an audio path that is using the same codec there is no
difference in performance as the codecs
do not do anything when both sides are the same.
On Mon, Aug 24, 2009 at 11:33 AM, Hound Dog d_ho...@ymail.com wrote:
Brian,
I set proxy_media to false ad now getting clean SDP
Your ACK message must not be valid (dialog matching or something else)
so every 1 call will generate 30 retries that are queued up in the sip
stack.
at 100cps you will be generating this problem 100 times per second and queue
up countless unfinished dialogs thus
eating up the cpu.
On Mon,
Actually I did that and it worked fine. I had the problem the SECOND time I
run FS and freepbx. And (@Brian) mod_sofia was loaded but sip_profiles were
not
On Sun, Aug 16, 2009 at 16:04, Carlos Talbot carlos.tal...@gmail.comwrote:
When you configure FreePBX for the first time it wipes out the
actually I meant the SECOND time I *RAN* FS and freepbx.
On Mon, Aug 24, 2009 at 20:24, Raffaele P. Guidi raffaele.p.gu...@gmail.com
wrote:
Actually I did that and it worked fine. I had the problem the SECOND time I
run FS and freepbx. And (@Brian) mod_sofia was loaded but sip_profiles were
If you installed FreePBX then it would be that softwares job to manage
the sofia profiles... wouldn't it?
/b
On Aug 24, 2009, at 1:24 PM, Raffaele P. Guidi wrote:
Actually I did that and it worked fine. I had the problem the SECOND
time I run FS and freepbx. And (@Brian) mod_sofia was
so problem solved , Thanks you anthony and Brian
From: Anthony Minessale anthony.miness...@gmail.com
To: freeswitch-users@lists.freeswitch.org
Sent: Monday, August 24, 2009 7:58:47 PM
Subject: Re: [Freeswitch-users] topology hiding leaking information in SDP
if
Hi Anthony,
I'm aware it is generating 30 retries per a call and this is killing me ...
I lost my entire working day to figure out what is missing in the damn ACK
message SIPp is sending back... ACK looks quite ok to me.
pls can you help ?
freeswi...@l01sipindir1 recv 573 bytes from
That is the remote sdp, not the local sdp. They are sending ptime 20,
not us. Are they actually sending 20 ms packets or are they sending 30?
MIke
On Aug 23, 2009, at 4:20 PM, Peter P GMX wrote:
Hello Anthony,
I set p...@30i,p...@30i and I can see in the logs that PCMA is used.
However
In your scenario you need to add [peer_tag_param] at the end of the to
on the Ack.
/b
On Aug 24, 2009, at 2:03 PM, Tihomir Culjaga wrote:
recv 383 bytes from udp/[10.4.4.252]:5060 at 16:44:26.535809:
Are you trying to test everything on the same machine?
/b
On Aug 24, 2009, at 2:24 PM, Jerry Richards wrote:
Hello All,
I am a Freeswitch Newbie and bringing up Freeswitch on my Windows XP
machine
for the first time using the Getting Started Guide. I can register
three
lines (1000,
Hello All,
I am a Freeswitch Newbie and bringing up Freeswitch on my Windows XP machine
for the first time using the Getting Started Guide. I can register three
lines (1000, 1001, and 1002), but when I attempt to call one phone to the
other I hear the operator say:
The person at extension 1000
Do you have an answer in the dialplan for that extension? Also, check
out the ignore_early_media variable.
Mike
On Aug 23, 2009, at 2:21 AM, Vassil Panayotov wrote:
Hi,
I managed to get our A500 running with FreeSWITCH 1.0.4 stable using
wanpipe 3.4.4 drivers. But now I have another
Every page on the wiki should be editable. If you don't already have
an account, go to:
http://wiki.freeswitch.org/index.php?title=Special:UserLogintype=signup
Mike
On Aug 22, 2009, at 12:42 PM, Lars Zeb wrote:
I think there’s something wrong with the script at
mr fritz is lying somewhere
get a pcap of the traffic from fritz to FS and look at the size of the audio
packets
if they are 160(172 with headers) bytes then it's 20ms if it's 240 (252)
then it's 30ms
if it's saying 20 but it means 30 you should leave the last change in place
and also add in
Yes. This a stand-alone Windows XP machine.
Jerry
-Original Message-
From: Brian West [mailto:br...@freeswitch.org]
Sent: Monday, August 24, 2009 12:33 PM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Cannot create outgoing channel type
[error]cause:
Make sure your x-lite is not on port 5060
/b
On Aug 24, 2009, at 3:25 PM, Jerry Richards wrote:
Yes. This a stand-alone Windows XP machine.
Jerry
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Hello Brian,
it doesn't work .. tried this today as well:
freeswi...@l01sipindir1 recv 573 bytes from udp/[10.4.4.252]:5060 at
20:28:09.367300:
INVITE
I just installed FS on a fresh Centos 5.3 install, everything went perfect. I
loaded x0lite up on a couple of hosts and they registered and worked fine. Then
I moved the Centos server to a new subnet on out network that we have tons of
boxes on and have all the routes already configured on and
ACK sip:[servi...@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: sipp sip:s...@[local_ip]:[local_port];tag=[call_number]
To: sut sip:[servi...@[remote_ip]:[remote_port][peer_tag_param]
Call-ID: [call_id]
CSeq: 1 ACK
Contact:
Hi Tihomir -
I'm no SIP guru, but the things which look suspicious about the ACK to
me are:
- Via header - different branch
- Contact header - differs from INVITE
--Dave
Hi Anthony,
I'm aware it is generating 30 retries per a call and this is killing
me ...
I lost my entire working day
Did you happen to hard code an IP in your vars.xml?
/b
On Aug 24, 2009, at 3:42 PM, Mike Peace wrote:
I just installed FS on a fresh Centos 5.3 install, everything went
perfect. I loaded x0lite up on a couple of hosts and they registered
and worked fine. Then I moved the Centos server to
Hello All,
I'm having an issue with g.729 pass-thru. On an older 1.0.3 install it's all
good, but on a newer machine, more cores, etc. the quality of voice on g.729
is very poor. Anyone here dealt with something like this?
With all other codecs are perfect, even in pass-thru...
Thanks in
On 25/08/09 2:05 AM, Anthony Minessale wrote:
I updated testclient.c so you can see how now.
Awesome awesome awesome!
Thanks heaps man - everybody here has been great!
--
Cheers,
Matt Riddell
Director
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I haven't changed any of the conf files.
Mike Peace
Network Analyst
EDCO, The Document People
Direct 417-447-3367
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Brian West
Sent: Monday, August 24, 2009 3:46 PM
To:
Registration error: 408-Request Timeout
Mike Peace
Network Analyst
EDCO, The Document People
Direct 417-447-3367
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Michael
Collins
Sent: Monday, August 24, 2009 4:18 PM
To:
oh yeah, it's only the port they let you pick
it looks like they don't let you pick ip in the abyss code, it would require
an intrusive patch into that depend lib
to allow you to set that.
On Mon, Aug 24, 2009 at 4:17 PM, Peter P GMX prometheus...@gmx.net wrote:
Any clue which one? I could
It does the same thing, does anything get set during the install that would
remember or cache the old network settings? I can access anything from the FS
server on any of several networks and vice-versa but the SIP will not register,
again no firewalls are upon any of the test hosts.
Doesn't
This is what I was asking! :D When the installer finished it started the
whole thing and everything got loaded fine, but when I restarted my system
it didn't (and did not anymore). Well, I will try to install everything from
scratch again and see...
On Mon, Aug 24, 2009 at 20:30, Brian West
Hello Brian, Dave
Still nothing... i've changed ip_addresses (remote_ip, local_ip) and changed
branch within ACK message to meet INVITE's onebut it is still not
enough...
Also i checked RFC and this is how should it be ... (ACK without contact
taking care to have correct TAGs and branch)...
What is your exact sipp scenerio file and dialplan to this point after the
changes you were suggested to use.
please send both.
If we have to stop what we are doing to prove this works are you prepared to
offer your soul to help
document and other project maintenance?
--
Anthony Minessale II
sipp_cmd: sipp -sn uac 10.4.4.251 -sf uac_redirect.xml -s
30003016094191500 -trace_err -r 1 -rp 1000 -trace_msg -inf test.txt -m 1 -l
4000
scenario file: uac_redirect.xml
FS dialplan: public.xml
SIP trace: trace.log
Here it is... sorry for not including at first...
On Mon, Aug 24, 2009 at 2:49 PM, Mike Peace mpe...@edcogroupinc.com wrote:
It does the same thing, does anything get set during the install that
would remember or cache the old network settings? I can access anything from
the FS server on any of several networks and vice-versa but the SIP
... documentation hate that :)) ... but thats my life actually... thats what
i do for living :)
of course i can go forward and document a lot of things... a lot of
docummentation is pending to completing the project i currently have using
FS.
well ... my soul? .. is it really necessary?
T.
On
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch-3]
shouldn't that be
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
On Mon, Aug 24, 2009 at 5:48 PM, Tihomir Culjaga tculj...@gmail.com wrote:
... documentation hate that :)) ... but thats my life actually...
this is original: Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-7085-1-0
http://sipp.sourceforge.net/doc3.0/reference.html
*[branch]* - Provide a branch value which is a concatenation of magic cookie
(z9hG4bK) + call number + message index in scenario.
An offset (like [branch-N]) can be appended
Is there any way to make FS complain about what header is wrong ?
T.
On Tue, Aug 25, 2009 at 1:14 AM, Tihomir Culjaga tculj...@gmail.com wrote:
this is original: Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-7085-1-0
http://sipp.sourceforge.net/doc3.0/reference.html
*[branch]* - Provide a
Hi Raymond,
I'm not planning to have more than 10 concurrent calls on this
devices, but I'm also curious as you about how many calls can it handle.
When I get to that point I will post the test results on this list.
Regards,
Rogelio
On Aug 23, 2009, at 8:23 PM, Raymond Chandler wrote:
On Tue, Aug 25, 2009 at 7:31 AM, Tihomir Culjagatculj...@gmail.com wrote:
sipp_cmd: sipp -sn uac 10.4.4.251 -sf uac_redirect.xml -s
30003016094191500 -trace_err -r 1 -rp 1000 -trace_msg -inf test.txt -m 1 -l
4000
scenario file: uac_redirect.xml
FS dialplan: public.xml
SIP
On Tue, Aug 25, 2009 at 10:52 AM, mayamatakeshimayamatake...@gmail.com wrote:
On Tue, Aug 25, 2009 at 7:31 AM, Tihomir Culjagatculj...@gmail.com wrote:
sipp_cmd: sipp -sn uac 10.4.4.251 -sf uac_redirect.xml -s
30003016094191500 -trace_err -r 1 -rp 1000 -trace_msg -inf test.txt -m 1 -l
Windows installer does not work for me.
I've reinstalled various times, same results.
I can correctly create a number, but when I try to create a device for
that number, it tells me that cannot locate the device, and the
password for vicemail will be invalid.
After that, it begins to give the
Hi -
Does anyone know how to compile freeswitch for a 64-bit architecture
on mac os x?
MySQL on MAC OS X only comes pre-compiled for 64-bit. I compiled
luasql against this libmysqlclient.dylib. Now when I call lua from
freeswitch I get:
2009-08-24 21:06:44.370439 [ERR] mod_lua.cpp:182 error
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