need to see the whole
dialog to see what is wrong... I tested with Polycom, Snom and Aastra.
Are you doing proxy media or anything like that?
/b
On Dec 29, 2009, at 1:14 AM, Lei Tang wrote:
Hi, I think hold function in trunk 16055 is broken, I have also tried some
old trunks, it's ok
+debug log and put
it on pastebin.
What phone are you using?
/b
On Dec 29, 2009, at 1:14 AM, Lei Tang wrote:
Hi, I think hold function in trunk 16055 is broken, I have also tried some
old trunks, it's ok in freeswitch 1.0.4.
The problem is, when send reponse for re-invite request, fs
as is.
/b
On Dec 29, 2009, at 8:53 AM, Lei Tang wrote:
Hi Brian, thanks for your help, I am using FS in proxy media mode. the
sip agent I'm using is x-lite and wxCommunicator.
I will test if trunk 16055 work when I set proxy media mode to false
tomorrow
Btw, in the same scenario, FS 1.0.4 works fine.
2009/12/29 Lei Tang lei.tl...@gmail.com
Hi Brian, I don't think so, I have debuged fs, If I'm not wrong, following
code in sofia.c send the 200ok response
sofia.c
function sofia_handle_sip_i_state
.
switch(ss_state
Hi, I think hold function in trunk 16055 is broken, I have also tried some
old trunks, it's ok in freeswitch 1.0.4.
The problem is, when send reponse for re-invite request, fs didn't send any
sdp content.
This problem is easy to reproduce, just call to fs, and press hold button,
Follow are sip
Hi all, I have found the cause of this problem. It due to some code in a
library I loaded into Fs, it set SIGPIPE handler, the handler seemed to be
invalid when SIGPIPE is fired, so FS is broken.
address, so,
2009/12/25 Lei Tang lei.tl...@gmail.com
BTW my is environment
[r...@localhost bin
/lib/jvm/java-1.4.2-gcj-1.4.2.0/jre --with-cpu=generic
--host=i386-redhat-linux
线程模型:posix
gcc 版本 4.1.2 20080704 (Red Hat 4.1.2-46)
2009/12/25 Lei Tang lei.tl...@gmail.com
Hi all and merry holiday, I have encounter fs core dump many times when I
exit fs_cli, I'm using the fs 1.0.5pre9.
I can
Hi all and merry holiday, I have encounter fs core dump many times when I
exit fs_cli, I'm using the fs 1.0.5pre9.
I can reproduce this fault by follow steps
1.launch fs with console
2.press ctrl+z to ext from fs console
3.run fs_cli (from local)
4.press ctrl+z to exit fs_cli (or type /bye)
5.fs
Hi All
I have a FS cluster behind two OpenSIPS proxy, the incoming calls is load
balance and failover to FS cluster by OpenSips, It works well.
The problem is, the outbound calls from FS must also route throw then
OpenSIPS servers. So, does FS servers can loadbalance the outbound calls
between
Hi everyone, I'm setting up FS cluster In my application, I plan to use
two FS server as front and four FS as backend, the incoming calls first
send to the front FS, then the front FS forward the call to backend FS
server by return 302 to invite message. The front FS need to known the
you can do this in follow steps:
1.edit default.xml diaplan config file in your fs config
directory(FS/conf/dialplan/default.xml), and section
extension name=ivr_demo2
condition field=destination_number expression=^\*114$
action application=lua data=../ivr/test.lua/
Hi, I'm running a ivr script on FS, the call is from a softswitch to extenal
sip endpoint of FS.
I added two dialplan in public dialplan xml file. as flow:
extension name=ivr_demo2
condition field=destination_number expression=^8$
action application=lua data=../ivr/test.lua/
Hi all, I'm testing a FS server using sipp, I found that sipp only show the
retrans of sip packet, Does someone known is there a tool to test FS rtp
packet lost rate in high concurrent call env?
--
Lei.Tang
lei.tl...@gmail.com
___
FreeSWITCH-users
- Implementing RTCP to identify lost packets
- Commercial hardware/software
If FreeSWITCH, your machine, or your network are pushed to the max and
falling apart you're most likely going to see audio problems on your
single (captured) call.
On Wed, Nov 11, 2009 at 8:43 AM, Lei Tang lei.tl...@gmail.com
As I opinion, it's not necessary write ivr script for each student. A
static ivr script load question and response dynamic is what you need.
2009/11/11 Malay Thakershi malay.thaker...@continuityhealth.com
Hello. I am very new to FreeSwitch, Telephony and IVR.
My goal is to prepare a
Congratulations!
2009/11/8 Seven Du dujinf...@gmail.com
ALL,
FreeSWITCH-CN is a non-official, non-profit Chinese community.
There was some arguments of language specified sites vs. a central site,
freeswitch.org, on this list. However, facts are that people would like to
find information
Hi all, I'm writing lua ivr scirpt, Does some known how to get digitals and
stop play when speak tts? Just like session:playAndGetDigits do. Thanks
lots!
Best Regards!
--
Lei.Tang
lei.tl...@gmail.com
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FreeSWITCH-users mailing list
Hi all,
The problem is solved. I ask the softswitch to send only sdp in INVITE
message, then It works.
I think sofia doesn't support multipart content currently. is it right?
2009/11/2 Lei Tang lei.tl...@gmail.com
Hi Daniel.
Sure. pls email me to tl...@hotmail.com.
2009/11/2 Zeng, Qinglan
Hi all, I get a 415 Unsupported Media Type when FS receiving call from a
softswitch. I captured some packets, It seems that the softswitch use SIP-I
protocol, does FS can handle SIP-I message?
===here is the invite messagefrom softswitch
INVITE sip:xx...@:5060;user=phone SIP/2.0
Contact:
)
nta_incoming_timer: 0/0 resent, 0/0 tout, 1/2 term, 1/2 free
nta: timer set next to 3859 ms
nta: timer I fired, terminate 415 response
incoming_reclaim_all(, , 02E2FEB8)
nta_incoming_timer: 0/0 resent, 0/0 tout, 1/1 term, 1/1 free
nta: timer not set
2009/11/2 Lei Tang lei.tl...@gmail.com
Hi all
Hi noob and Michael, thanks for your answers, I'll try to use
mod_xml_curl.
Hi Henry, http://wiki.freeswitch.org/wiki/Mod_xml_curl has mentioned, FS
will post a request to webserver when it get a registration request. you can
refer to the doc for more detail.
Lei.Tang
lei.tl...@gmail.com
Hi all, I run FS on a machine with two net interface, each interface has a
ip addr, one of the them connect to public network(has ip addr A), the
other connect to a private network(has ip addr B), FS server as a SIP
server for public through A, all outbound call will bridge to a softswitch
in
Thanks Eliot, It works.
2009/10/27 Eliot Gable
egable+freeswi...@gmail.comegable%2bfreeswi...@gmail.com
Try setting ext-rtp-ip and ext-sip-ip on both profiles.
On Tue, Oct 27, 2009 at 4:49 AM, Lei Tang lei.tl...@gmail.com wrote:
Hi all, I run FS on a machine with two net interface, each
no need to set the ext-*-ip equiv.
/b
On Oct 27, 2009, at 8:14 PM, Lei Tang wrote:
Thanks Eliot, It works.
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http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
Hi All:
I'm a newbie to FS. I'm using FS as a sbc and have about 2 user
account . Does somebody can tell me how to make FS load use account
information from a database such as mssql or mysql? Could you give me a
sample configuration file?
Thanks a lots.
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