February 2011 - FreeSWITCH-users - FreeSWITCH Mailing Lists

txfax and multiple files
by Madovsky 17 Feb '11

17 Feb '11

What is the right syntax to send multiple files with mod_spandsp txfax ? thanks

2 2

mod_xml_curl response debug
by David Wafula 17 Feb '11

17 Feb '11

Am using mod_xml_curl to fetch dialplans from a web app. However, am running into: mod_xml_curl.c:301 Error Parsing Result! I have checked my xml responses, they look fine, but obviously something is wrong somewhere. So how do i get to see the response freeswitch gets in the logs, to help me narrow down the problem? -- David Wafula

2 2

Possible to communicate with FS using event_socket without ESL?
by Bert Jones 17 Feb '11

17 Feb '11

Hello, Am I correct in assuming that an app can talk directly to FS using event_socket without the use of the ESL? I am attempting to do this, but having trouble authenticating. I wanted to make sure I was not overlooking something basis in my approach. I have successfully loaded FS on a windows XP server and can register phones and make calls. Further I can talk to the server, authenticate and make calls using telnet, but when attempting to do the same via an app written in .net and using a socket, I am unable to authenticate. Running my app using a tcpSocket, I receive the "Content-Type: auth/request" upon connection of the socket, but when sending "auth password\n\n" it appears that the connection times out after about 10 seconds and I receive: "Content-Type: text/disconnect-notice" & vbLf & "Content-Length: 67". mod_event_socket is enabled in modules.conf.xml . Is there something obvious that I am missing conceptually, or should this work? Thanks!

6 11

Re: [Freeswitch-users] g729 packets skipped in passthrough mode
by Javier Gallart 17 Feb '11

17 Feb '11

> > Hello > my mistake... I had written data="RTP_BUG_IGNORE_MARK_BIT" instead of value="RTP_BUG_IGNORE_MARK_BIT". It's working now. Keep this in mind if you ever happen to put a Huawei SBC in from of a FS! Thanks Javier > > ---------- Forwarded message ---------- > From: Javier Gallart <jgallartm(a)gmail.com> > To: freeswitch-users(a)lists.freeswitch.org > Date: Wed, 16 Feb 2011 13:28:35 +0100 > Subject: Re: [Freeswitch-users] g729 packets skipped in passthrough mode > Hello, > > I've looked deeper into this issue and I might have hit a known problem: > > RTP_BUG_IGNORE_MARK_BIT = (1 << 2) > > /* > A Huawei SBC has been discovered that sends the mark bit on every single > RTP packet. > Since this causes the RTP stack to flush it's buffers, it horribly messes > up the timing on the channel. > > This flag will do nothing when an inbound packet contains the mark bit. > > */ > > The carrier is actually a Huawei SBC, and the problem I reported only > happens when it sends the marker bit = true in all the rtp packets. I've > tried > <param name="auto-rtp-bugs" data="RTP_BUG_IGNORE_MARK_BIT"/> in the profile > but the problem is still there, is this the right way to override this > nonsense from the Huawei? > > Thanks > > Javier > >> >> ---------- Forwarded message ---------- >> From: Anthony Minessale <anthony.minessale(a)gmail.com> >> To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> >> Date: Tue, 15 Feb 2011 16:38:35 -0600 >> Subject: Re: [Freeswitch-users] FreeSWITCH-users Digest, Vol 56, Issue 177 >> Well that's all I could think of: >> >> passthrough is passthrough FS never modifies anything. >> you need to look harder at sip trace, pcaps or other diagnostics for >> some misconfiguration. >> many people use this daily. >> >> >> On Tue, Feb 15, 2011 at 1:12 AM, Javier Gallart <jgallartm(a)gmail.com> >> wrote: >> > >> > Anthony, thanks for the tip. I haven't seen any change though. I should >> have >> > mentioned that I'm focusing in the rtp stream coming form the callee to >> the >> > caller. The problem is that FS relays to the caller only half of the >> packets >> > received from the callee. The packets arrive at FS at a rate of 1 packet >> > every 20 ms, and with a payload of 20 bytes each. Packets from FS to the >> > caller are sent every 40 ms with a payload of 20 bytes, thus skipping >> half >> > of the information. >> > >> > Thanks >> >> >> >> ---------- Forwarded message ---------- >> >> From: Anthony Minessale <anthony.minessale(a)gmail.com> >> >> To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> >> >> Date: Mon, 14 Feb 2011 10:38:57 -0600 >> >> Subject: Re: [Freeswitch-users] g729 packets skipped in passthrough >> mode >> >> I bet its 20ms vs 30 >> >> set passthru_ptime_mismatch to true either in vars.xml or in your >> >> dialplan both legs through export on the a leg or set in the a leg and >> >> in the {} on b. >> >> >> >> >> >> On Mon, Feb 14, 2011 at 10:28 AM, Brian West <brian(a)freeswitch.org> >> wrote: >> >> > What exactly is the problem? I see no issue here can you elaborate >> on >> >> > what you're seeing? >> >> > >> >> > /b >> >> > >> >> > On Feb 14, 2011, at 4:48 AM, Javier Gallart wrote: >> >> > >> >> >> I've tried to explicitly set ptime at 20ms at switch.conf -although >> >> >> it's not necessary afaik-. Has any one experienced this same issue? >> >> >> >> >> >> Thanks in advance.. >> >> >> >> >> >> Regards >> >> > >> >> > >> >> > >

3 2

Can we Intercept a sleeping channel?
by mayamatakeshi 17 Feb '11

17 Feb '11

To hold a call to be picked up using intercept, I've tried this with the intention of having the call sent to voicemail if no one does the pickup: <action application="sleep" data="20000"/> <action application="answer"/> <action application="voicemail" data="default user(a)domain.com"/> However, when I use application intercept, sleep is not aborted and we have no audio until it finishes. I was expecting intercept would abort the sleep application the same way as it does for example for application playback. I believe this can be solved by using application park and sched_transfer. But I thought I could ask if there would be a way of doing it without splitting the dialplan entities and keep it simple. br, takeshi

1 0

IVR application
by Jason b.a 17 Feb '11

17 Feb '11

Hey guys, i need to implement an IVR application, but i am wondering if it is required to use a SIP SERVER between my IVR application and FREESWITCH. thank you

3 3

OSTAG Facebook page
by Brian West 16 Feb '11

16 Feb '11

http://www.facebook.com/pages/Open-Source-Telephony-Advancement-Group-Inc/1… Please join the OSTAG facebook community... it has taken a long time to get non-profit status so please help out and donate if you can... get involved. Thanks, Brian

2 1

Core dump error
by rex.alex 16 Feb '11

16 Feb '11

Hi I was using the FreeSWITCH 1.0.6 for quite some time in production and was very stable and recently I had upgraded to latest git to use the mod_spandsp module. From then on I used to get core dump issue sometimes and hence yesterday I did a git pull && make current on source. But today, till now FreeSWITCH has been core dumped for 5 times, could not find the reason. I have pastebin the core trace in http://pastebin.freeswitch.org/15402 Version : FreeSWITCH Version 1.0.head (git-722b2ce 2011-02-15 19-01-45 -0600) OS : Cent OS 5.5 Please help me to resolve the same. Thanks, Rex -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Core-dump-error-tp6033400p603… Sent from the freeswitch-users mailing list archive at Nabble.com.

2 1

Benchmarking
by Wagner 16 Feb '11

16 Feb '11

Hello, I'd like to know how could I benchmark a FreeSwitch server? Is there any tool to stress it? Thanks

2 1

FreeSWITCH Conference Call Today
by Michael Collins 16 Feb '11

16 Feb '11

Hey folks, We are having a "get caught up with the latest in FreeSWITCH" session today. Here's the agenda: http://wiki.freeswitch.org/wiki/FS_weekly_2011_02_16 I will be reviewing a lot of the stuff that went into FreeSWITCH in the month of January. If you are familiar with some of the new things please feel free to add your input to the call today and/or update the wiki page. For example, if you are familiar with the mod_snmp or mod_fsk we would definitely welcome your observations. Talk to you soon, Michael

1 0