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

because > in proxy media they are passed 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 wxCommunicat

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

Btw, in the same scenario, FS 1.0.4 works fine. 2009/12/29 Lei Tang > 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 >..

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

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

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

he 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 &

[Freeswitch-users] Hold is broken in trunk 16055

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 tr

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

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 > BTW my is environment > > [r...@localhost bin]# uname -

[Freeswitch-users] fs core dump after fs_cli disconnected

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 co

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

/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 > 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 reprod

Re: [Freeswitch-users] FS doesn't update rtp port when sip UPDATE message changed the rtp port

Thanks Michael, sorry for my mistake, I'm using FS 1.0.5pre9, I'll try the lastest svn trunk. 2009/12/23 Michael Jerris > There is no such thing as freeswitch 1.5. Have you tried latest svn trunk > to see if this behavior is the same? > > Mike > > > On Dec 23, 20

[Freeswitch-users] FS doesn't update rtp port when sip UPDATE message changed the rtp port

Hi all, I'm using FS 1.5, doesn't somebody known something about this problem? My scenario is : A(FreeSwitch) B --INVITE ---> <100 Tring <180 Ring with sdp m=audio 55066 RTP/AVP 0 120 c=IN IP4 10.36.143.76

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

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 t

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

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 2. edit your ivr script, your can refer to http://wiki.freeswitch.org/wiki/Mod_lua for how to write ivr script in lua. 3.

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

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 backend

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

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: Every thing is ok when call to number 8. but when I call the second number "*114", fs

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

- 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, 2

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

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 maili

Re: [Freeswitch-users] Help with dynamic IVR

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 > Hello. I am very new to FreeSwitch, Telephony and IVR. > > > > My goal is to prepare a student assessment IVR system as a

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

Congratulations！ 2009/11/8 Seven Du > 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 in their nati

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

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 FreeSWITCH-users

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

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 > Hi Daniel. > Sure. pls email me to tl...@hotmail.com. > > 2009/11/2 Zeng

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

) 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 > Hi all, I get a &

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

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: M

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

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://l

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

Thanks Eliot, It works. 2009/10/27 Eliot Gable > > Try setting ext-rtp-ip and ext-sip-ip on both profiles. > > On Tue, Oct 27, 2009 at 4:49 AM, Lei Tang wrote: > > Hi all, I run FS on a machine with two net interface, each interface has > a > > ip addr, one of

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

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 priv

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

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 load user account from databse ?

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. ___