- FreeSWITCH-dev - FreeSWITCH Mailing Lists

FreeSWITCH Conference Call Today - Safi Systems IVR Builder
by Michael Collins 27 Apr '11

27 Apr '11

Hello all! We are happy to have Zac Wolfe from Safi Systems joining us today to talk about the IVR building system they have created. There is a gotomeeting.comlink on the agenda page here: http://wiki.freeswitch.org/wiki/FS_weekly_2011_04_27 We look forward to this discussion! -Michael

1 0

One way audio problem from B-leg of the call on Session Refresh and HOLD if its a dynamic payload type
by Jyotshna Cherukuri 26 Apr '11

26 Apr '11

Hi , I am working with the latest revision of Freeswitch and I am having one way audio problem using SILK/8000 codec on B-leg of the call after Session Refresh or on HOLD. This is due to the fact that when FS sends an INVITE out to B-leg it sends 98 in its SDP offer and B-leg responds back with same code but with payload type "120" [ Offer ] v=0 o=FreeSWITCH 1303773966 1303773967 IN IP4 192.168.2.10 s=FreeSWITCH c=IN IP4 192.168.2.10 t=0 0 m=audio 33766 RTP/AVP 98 9 101 a=rtpmap:98 SILK/8000 a=fmtp:98 useinbandfec=1;usedtx=0 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 [ Answer ] v=0 o=- 3512815772 3512815773 IN IP4 192.168.4.121 s=pjmedia c=IN IP4 192.168.4.121 t=0 0 a=X-nat:0 m=audio 4002 RTP/AVP 120 96 a=rtcp:4003 IN IP4 192.168.4.121 a=rtpmap:120 silk/8000 a=fmtp:120 useinbandfec=1;usedtx=0 a=sendrecv a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-15 Freeswitch handles this properly on the initial offer/answer as its using this patch (tell rtp stack about what remote payload type to expect when the receiving end follows the stupid SHOULD as WONT and sends a different dynamic payload number than the one in the offer) After one min into the call because Session Timers are enabled Freeswitch sends a Session Refresh with payload type now setting to "120" [Refresh Offer] v=0 o=FreeSWITCH 1303773966 1303773967 IN IP4 192.168.2.10 s=FreeSWITCH c=IN IP4 192.168.2.10 t=0 0 m=audio 33766 RTP/AVP 120 96 9 a=rtpmap:120 SILK/8000 a=fmtp:120 useinbandfec=1;usedtx=0 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 a=silenceSupp:off - - - - a=ptime:20 The remote end then starts sending RTP packets with payload number "120" in its RTP header and FS stops processing these packets and as a result is resulting in one-way audio issue. Any help is appreciated. Thanks in advance Regards Jyotshna P.S : The issue starts even when the remote end presses" HOLD" as it sends INVITE on hold with "120" and FS responds back with "120" in its answer.

3 2

Help: how to disable dynamic RTP mapping?
by Oleg Khovayko 25 Apr '11

25 Apr '11

Hi, My FreeSWITCH machine is located behind the NAT-device. When I receive incoming call, I see 10 secs delay between I picked up handset and voice connection establishing. During these 10s, caller hears long beeps, as same as "ringing" signal. I investigated problem, and found: after recipient pick up handset, FreeSWITCH tried dynamically map two UDP ports for RTP-connections. I use Fios router from Verizon, and I assume, it slowly adding dynamic port mapping. It adds one RTP port per 5s. So, when FS dynamically maps both ports, it loose 10 seconds (5+5). Log follows. I have added static rule into router -- map UDP ports 20000~25000 to FS box. And, in the FS_box, I added into file ./autoload_configs/switch.conf.xml <param name="rtp-start-port" value="20000"/> <param name="rtp-end-port" value="25000"/> But, anyway, FS tries dynamically assign ports for each income call, and lost 10s every time. Please suggest, how to disable dynamic port mapping, and use static port range mapping (20000-25000) only? I assume, this will increase speedup of my system. Thanks, Oleg 2011-04-22 18:14:45.590723 [DEBUG] switch_channel.c:2813 (sofia/internal/sip:1001@192.168.1.151:5060) Callstate Change RINGING -> ACTIVE . . . . 2011-04-22 18:14:45.750984 [DEBUG] switch_core_session.c:1116 Send signal sofia/internal/sip:1000@192.168.1.152:5060 [BREAK] 2011-04-22 18:14:45.750984 [DEBUG] switch_core_session.c:1288 Session 57 (sofia/internal/sip:1000@192.168.1.152:5060) Locked, Waiting on external entities <5s delay> 2011-04-22 18:14:50.018621 [DEBUG] sofia_reg.c:1765 Changing expire time to 109 by request of proxy sip:callcentric.com 2011-04-22 18:14:50.209565 [DEBUG] switch_nat.c:502 mapped public port 20794 protocol UDP to localport 20794 2011-04-22 18:14:50.341416 [DEBUG] sofia_reg.c:1765 Changing expire time to 104 by request of proxy sip:callcentric.com <5s delay> 2011-04-22 18:14:55.213775 [DEBUG] switch_nat.c:502 mapped public port 20795 protocol UDP to localport 20795 2011-04-22 18:14:55.213775 [DEBUG] sofia_glue.c:3001 AUDIO RTP [sofia/external/2404760839(a)66.54.140.46] 192.168.1.5 port 20794 -> 204.11.192.22 port 60140 codec: 0 ms: 20 2011-04-22 18:14:55.213775 [DEBUG] switch_rtp.c:1623 Starting timer [soft] 160 bytes per 20ms

2 3

[article] High-performance Timing on Linux / Windows
by Tamas Jalsovszky 23 Apr '11

23 Apr '11

Hello, I've found an article about the topic, maybe you can find something in it: http://tdistler.com/2010/06/27/high-performance-timing-on-linux-windows If not, please ignore :) Regards, T.

4 3

How to use "say" command for a conference.
by Min Xie 21 Apr '11

21 Apr '11

All, We tried to do a "say" command to a conference but always got the following. What's the right console command line parameters to perform "conference say" or "conference saymember"? freeswitch@192.168.0.123@internal> conference myconference-name say "hello world" (say) Error! say <text> Thanks! Min Xie ________________________________ NOTICE: The information contained in this electronic mail transmission is intended by Convergys Corporation for the use of the named individual or entity to which it is directed and may contain information that is privileged or otherwise confidential. If you have received this electronic mail transmission in error, please delete it from your system without copying or forwarding it, and notify the sender of the error by reply email or by telephone (collect), so that the sender's address records can be corrected.

2 1

Noise heard when a user holds a conference in SRTP mode
by Ujjwal SIngh 21 Apr '11

21 Apr '11

Hi, I have FreeSwitch configured as a conference bridge, the sip_secure_media=true in dialplan and vars.xml. User A, B and C are in a conference, the media is established in SRTP mode, when any user presses hold, other members hear noise, while in RTP mode normal Music on Hold is heard. Can anybody please help what is going wrong in SRTP case. I have uploaded FreeSwitch log in pastebin below http://pastebin.freeswitch.org/16103 Thanks for the help. Regards, Ujjwal

2 3

FreeSWITCH Conference Call Today - VoIP Abuse/PBX Honeypot
by Michael Collins 20 Apr '11

20 Apr '11

Hello all! Our agenda is here: http://wiki.freeswitch.org/wiki/FS_weekly_2011_04_20 At about 2PM EDT (11AM PDT) J. Oquendo will join us to talk about PBX Honeypot and VoIP Abuse projects. We will still start the conference at 1PM EDT/10AM PDT and do our agenda items until J. joins us. Talk to you soon! -Michael

1 0

[Re] Implementing Forward-{All, Busy, NoAnswer}, ring policies and DND in FS
by Marc Olivier Chouinard 20 Apr '11

20 Apr '11

Hi Mathieu, I think it a great idea... it something I was thinking to go look for that server side forward feature of SIP supported by Polycom. Going to be hard to define something standard for all endpoint. The basic forward all,no answer(number of ring), busy are fine to do, but the ring-on-idle and blink-on-idle... seem specific to skinny. As far as I know, in sip you would send an alert with a silence ring tone. So not sure how we would want to implement that one. In case of SIP implementation, it probably a SIP XML msg we receive... So what do we do with it ? Generate a dummy call to the dialplan ? Or should the different endpoint generate an standard event, and we make a new module that monitor those event and does populate whatever is needed ? I'm trying to think long term also of other feature we might want to implement (SIP Conferencing monitoring support for example...) Moc

1 0

Proposal to add a Deprecated Log Level
by Marc Olivier Chouinard 20 Apr '11

20 Apr '11

Hi, I would like to propose adding a Deprecated Log Level to Freeswitch. This would serve as a special warning to user that some feature shouldn't be used and might be removed in a more visible way. Also I propose we use a number for every log we do. like DEPRECATED00001 (or whatever), and create wiki entry related to it. In the wiki, we would have to put the information the user need to change to use the current function. Also we can specify a due date for removal of the feature. We could also create a special macro that we can specify the unix time for the removal of the feature and make it display as critical error if the date is less than 1 month (or something). This would help to cleanup the code in the long term without having the adverse affect of creating surprises to user. Moc

1 0

Originated session callback.
by Mohammed Habib 18 Apr '11

18 Apr '11

I need help getting events from originated session. This is my lua script: function onInput_MainSession(s, type, obj) -- This one is working fine. freeswitch.consoleLog("info", "Callback with type " .. type .. "\n"); end function onInput_NewSession(s, type, obj) -- This one is never called. freeswitch.consoleLog("info", "Callback with type " .. type .. "\n"); end session:answer(); session:setInputCallback("onInput_MainSession"); session:sleep(200); session:execute("detect_speech", "unimrcp testgrammer trestgrammer"); newsession = freeswitch.Session("user/1002"); newsession:setInputCallback("onInput_NewSession"); newsession:sleep(200); newsession:execute("detect_speech", "unimrcp testgrammer trestgrammer"); while ((session:ready() == true) ) and (newsession:ready() == true) do -- Loop sleep(200); end I am unable to capture any of the new session events or dtmf. Please help. Thank you, Mohammed Habib

1 0