El Viernes, 28 de Noviembre de 2008, Brian West escribió:
Have the rfc?
RFC 4825 -- XML Configuration Access Protocol (XCAP)
RFC 4826 -- XML Formats for Representing Resource Lists
RFC 4827 -- XCAP Usage for Manipulating Presence Document Contents
RFC 5025 -- Presence Authorization Rules
I m trying to play some basics on FreeSwitch plus twinkle softphone. Following
error msg is displayed on the softphone:
Failed to create a udp socket on port 5060. Address already in use
following on freeswitch window:
X Error: BadWindow (invalid Window parameter) 3
Major opcode: 20
Minor
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hello,
This problem is solved. In fact it wasn't a freeswitch problem. I had to
delete the whole directory where freeswitch was installed. After a
reinstall everything works as expected.
Sorry for making trouble.
regards
helmut
Helmut Kuper
this shows that port 5060 is already used by another application,
i think you are running both Freeswitch and twinkle softphone on same
machine,
if its true then try to set different port number to use for softphone..
Faisal Maqsoodi wrote:
I m trying to play some basics on FreeSwitch plus
On Fri, Nov 28, 2008 at 8:45 AM, Faisal Maqsoodi
[EMAIL PROTECTED]wrote:
I m trying to play some basics on FreeSwitch plus twinkle softphone.
Following error msg is displayed on the softphone:
Failed to create a udp socket on port 5060. Address already in use
following on freeswitch window:
Hy!
There are two different FS behind the same NAT, and there were Reigstration
Failures about one or to times a day. The gateway status turned down, then I
got 503 error codes. Then I set up the ext-ip to STUN, as the wiki requests it.
Now I facing the next problem:
Start the call,
hi,
i wonder, if or what i do not understand how to do send_dtmf in the right way.
for example i want to send dtmf tones to my mobile mailbox the enter
the menu and do some changes to the settings. but whatever i try, it
does not work.
i tried:
sendmsg send_dtmf outbound_uuid 123#
sendmsg
so i would have to make a call with a phone to a specific dialplan? if
so, this would not be, what i whished (although it is nice to have the
option).
isn't there something, which can stream the voice of a given uuid? so
i could place a link in the html admin-area to spy an uuid and to hear
X-PRE-PROCESS cmd=set data=local_ip_v4=1.2.3.4/
/b
On Nov 28, 2008, at 2:22 AM, Carole O. wrote:
Hello,
I think I will keep first the first solution since I do not know yet
how to
reset a variable.
Thank you very much for your help.
Carole
Brian West-3 wrote:
You can reset what
I will bet you that the 183 has no port or the wrong port in the
contact.
/b
On Nov 28, 2008, at 8:20 AM, x y wrote:
then after 183 Session progress from port 1352, the PRACK package
goes to 5060 instead of 1352,
___
Freeswitch-users mailing
oh ok, I thought you meant something else. This one I got it but thanks for
keeping answering.
Carole
Brian West-3 wrote:
X-PRE-PROCESS cmd=set data=local_ip_v4=1.2.3.4/
/b
On Nov 28, 2008, at 2:22 AM, Carole O. wrote:
Hello,
I think I will keep first the first solution since I
The whole situation:
xxx.xxx.xxx.xxx:56956---INVITE---gt;yyy.yyy.yyy.yyy:5060
yyy.yyy.yyy.yyy:5060---100-Trying---gt;xxx.xxx.xxx.xxx:5060
yyy.yyy.yyy.yyy:5060---INVITE---gt;zzz.zzz.zzz.zzz:1352
zzz.zzz.zzz.zzz:1352---100-Trying---gt;yyy.yyy.yyy.yyy:5060
I don't really care where or to the packets come from but the actual
contents of the contact header in each packet.
/b
On Nov 28, 2008, at 9:50 AM, x y wrote:
The whole situation:
xxx.xxx.xxx.xxx:56956---INVITE---yyy.yyy.yyy.yyy:5060
yyy.yyy.yyy.yyy:5060---100-Trying---xxx.xxx.xxx.xxx:5060
I'm getting the following errors when trying to run the example in the wiki:
http://wiki.freeswitch.org/wiki/Mod_openmrcp
2008-11-28 09:59:54 [DEBUG] switch_core_session.c:435
switch_core_session_receive_message() Send signal sofia/internal/[EMAIL
PROTECTED] [BREAK]
2008-11-28 09:59:54 [DEBUG]
Hello,
I'm using event socket outbound, and have an issue where, after a bridge
ends and is terminated by Leg B, Leg A is also terminated. Here's the
call flow:
1. Call comes in (Leg A), session created, play welcome message.
2. From this session, originate and dial out using
Can you get me a pcap of this scenario? I can almost bet the contact
changes in the 183 which causes sofia to send the PRACK to the new
port which is the correct behavior. The other option is to turn off
100rel.
/b
On Nov 28, 2008, at 8:50 AM, x y wrote:
Cheers,
Viktor
I have a phone that is registered to FS but is no longer available
(Internet connection down, phone turned off, etc.). The registration
still exists in the sip_registrations table (not expired yet), but the
phone is not reachable on the network.
According to my dialplan, if the bridge to the
Try pre_answer before bridge.
/b
Sent from my iPhone
On Nov 28, 2008, at 3:03 PM, Gabriel Kuri [EMAIL PROTECTED] wrote:
I have a phone that is registered to FS but is no longer available
(Internet connection down, phone turned off, etc.). The registration
still exists in the
Hello,
I am using a GSM based endpoint connected to freeswitch that makes calls to
the PSTN via a SIP gateway (SBC). The SBC uses PCMU between itself and
freeswitch.
When I make an outgoing call from a GSM based device via freewsitch to the
PSTN via the SBC, everything works fine and audio
Yes, i m running both on the same machine. Now i changed it to port 5020 for
twinkle. But the error msg displayed is:
Sat 10:37:25
1001, registration failed: 503 Service Unavailable
What should i try?
--- On Fri, 11/28/08, shehzad p [EMAIL PROTECTED] wrote:
From: shehzad p [EMAIL PROTECTED]
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