Re: [Freeswitch-users] Hold is broken in trunk 16055

2009-12-29 Thread Lei Tang
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

Re: [Freeswitch-users] Hold is broken in trunk 16055

2009-12-29 Thread Lei Tang
+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

Re: [Freeswitch-users] Hold is broken in trunk 16055

2009-12-29 Thread Lei Tang
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

Re: [Freeswitch-users] Hold is broken in trunk 16055

2009-12-29 Thread Lei Tang
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

[Freeswitch-users] Hold is broken in trunk 16055

2009-12-28 Thread Lei Tang
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

Re: [Freeswitch-users] fs core dump after fs_cli disconnected

2009-12-26 Thread Lei Tang
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

Re: [Freeswitch-users] fs core dump after fs_cli disconnected

2009-12-25 Thread Lei Tang
/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

[Freeswitch-users] fs core dump after fs_cli disconnected

2009-12-25 Thread Lei Tang
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

[Freeswitch-users] how does FS failover or load balance outbound calls between tow proxy

2009-12-17 Thread Lei Tang
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

[Freeswitch-users] FS cluster and how to get sofia gateway health status?

2009-11-24 Thread Lei Tang
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

Re: [Freeswitch-users] How to run IVR application

2009-11-24 Thread Lei Tang
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/

[Freeswitch-users] hangup incoming call by Reason: Q.850; cause=1; text=Unallocated (unassigned) number

2009-11-12 Thread Lei Tang
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/

[Freeswitch-users] How to test FS rtp packet lost rate?

2009-11-11 Thread Lei Tang
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

Re: [Freeswitch-users] How to test FS rtp packet lost rate?

2009-11-11 Thread Lei Tang
- 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

Re: [Freeswitch-users] Help with dynamic IVR

2009-11-10 Thread Lei Tang
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

Re: [Freeswitch-users] Announce FreeSWITCH-CN - the Chinese community

2009-11-07 Thread Lei Tang
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

[Freeswitch-users] How to get digitals and stop play when speak tts? Just like session:playAndGetDigits do

2009-11-03 Thread Lei Tang
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 ___ FreeSWITCH-users mailing list

Re: [Freeswitch-users] Get error 415 Unsupported Media Type whenreceiving call from softswitch

2009-11-02 Thread Lei Tang
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

[Freeswitch-users] Get error 415 Unsupported Media Type when receiving call from softswitch

2009-11-01 Thread Lei Tang
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:

Re: [Freeswitch-users] Get error 415 Unsupported Media Type when receiving call from softswitch

2009-11-01 Thread Lei Tang
) 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

Re: [Freeswitch-users] How to load user account from databse ?

2009-10-27 Thread Lei Tang
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

[Freeswitch-users] how to config FS with two net interface?

2009-10-27 Thread Lei Tang
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

Re: [Freeswitch-users] how to config FS with two net interface?

2009-10-27 Thread Lei Tang
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

Re: [Freeswitch-users] how to config FS with two net interface?

2009-10-27 Thread Lei Tang
no need to set the ext-*-ip equiv. /b On Oct 27, 2009, at 8:14 PM, Lei Tang wrote: Thanks Eliot, It works. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users

[Freeswitch-users] How to load user account from databse ?

2009-10-26 Thread Lei Tang
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.