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application and media server
by Jason b.a 17 Feb '11

17 Feb '11

Hi , should i use sip server to connect the application server with the freeswitch (media server). client ---- application server --- sip server ---- freeswitch and is MSCML or VXML embedded in the sip message so the application server can control the freeswitch. thank you .

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non-preemption in play_and_get_digits
by abubacker 17 Feb '11

17 Feb '11

Is this possible to set the non preemption mode int the play_and_get_digits_application ? If yes then how ? -- Best regards, N.Abubacker , Associate system engineer , bk systems pvt ltd , Ph : 9144-43902701 Disclaimer: http://www.bksystems.co.in/email-policy

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IVR application
by Jason b.a 17 Feb '11

17 Feb '11

yeh i am using xlite , so my design will be xlite--IVR application--freeswitch but what about the signaling or control protocol used between app and Fs.

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Broad-voice configuration trouble.
by Victor Chukalovskiy 17 Feb '11

17 Feb '11

Hello, I can't get BroadVoice running. Currently I follow this page: http://wiki.freeswitch.org/wiki/Provider_Configuration:_Broadvoice and it registers Ok. However, outgoing calls fail with Broadvoice sending me "403 Forbidden". I do set absolute_codec_string to PCMU and I see that PCMU is indeed the only codec in my SDP. Any help is appreciated. Actually, can anyone share an INVITE packet that works with Broad-voice? I think I'll be able to solve it then. Thank you, Victor > Sorry for hijacking previous thread!

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premature disconnect on outbound call: destroy/unlink session from object
by Neil Patel 17 Feb '11

17 Feb '11

Hi All, I am running into an intermittent problem with outbound calling. I originate calls (using FreeTDM to Sangoma hardware) from a lua script, and every so often (10-50% of the time) the call disconnects just as the endpoints answers the call. I am executing these calls identically and to the same phone number. Here is CLI output for a call that doesn't disconnect: http://pastebin.freeswitch.org/15409 And here is an example of one that does upon answering on the endpoint: http://pastebin.freeswitch.org/15408 I noticed that one difference is that in the disconnected call there is the following output: 2011-02-17 23:11:48.747752 [DEBUG] switch_core_state_machine.c:204 FreeTDM/2:1/09586550654 Standard SOFT_EXECUTE 2011-02-17 23:11:48.747752 [DEBUG] switch_core_state_machine.c:372 (FreeTDM/2:1/09586550654) State SOFT_EXECUTE going to sleep *2011-02-17 23:11:48.747752 [DEBUG] switch_cpp.cpp:981 FreeTDM/2:1/09586550654 destroy/unlink session from object* 2011-02-17 23:11:48.747752 [DEBUG] switch_channel.c:2535 (FreeTDM/2:1/09586550654) Callstate Change ACTIVE -> HANGUP I'm wondering what's triggering that destroy/unlink. I've taken this issue up with Sangoma support, they are saying that there is nothing from the HW side that would trigger this, though freeTDM may possible be passing back some variables in a way that causes this and subsequent callstate change. In case it's useful, pcaps for the normal,connected call and disconnected fault call are attached. Thanks in advance, Neil

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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

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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

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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!

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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 >> >> > >> >> > >> >> > >

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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

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