August 2010 - freeswitch-dev@lists.freeswitch.org

[Freeswitch-dev] FS complained on RTP packet.
by Johny Kadarisman Kwan 27 Aug '10

27 Aug '10

I'm trying to convert a proprietary audio stream into sip/rtp compatible. At this point, i'm able to pass sip negotiation with FS, and trying to stream PCMU/8000 codec. But something not right on my end, and FS complained about the ptime settings. I'm sending 320 bytes ulaw payload on every rtp packet. Is that the right amount? appreciate if anyone could point some info on these topics. Thanks, JK =============== *2010-08-25 16:36:58.142328 [WARNING] mod_sofia.c:1013 We were told to use ptime 20 but what they meant to say was 40* *This issue has so far been identified to happen on the following broken platforms/devices:* *Linksys/Sipura aka Cisco* *ShoreTel* *Sonus/L3* *We will try to fix it but some of the devices on this list are so broken,* *who knows what will happen..* *===================*

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[Freeswitch-dev] Module written in C++
by Farhan Husain 27 Aug '10

27 Aug '10

I am trying to write a module having multiple C++ source files. When I run make, I find that only the C++ file having the name same as the module name is being compiled and all others do not produce any object file. As a result when I run FS I get linker error. Can anyone tell me how to get them compiled too? Thanks, Farhan

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[Freeswitch-dev] Freeswitch communication delay after three hours
by Achim Stamm 26 Aug '10

26 Aug '10

Hello! I have following problem: My current session is connected to a thirdParty Application by using an own FreeSwitch application "DoingReadAndWriteFrames". The communication between FreeSwitch Session and thirdParty Application is done by reading and writing frames in a while loop. At start of FreeSwitch Session the communication works without delay. After three or more hours i get a communication delay of one or more seconds. If i hang up and make a new call (a new freeswitch session), than the delay disappears, but after several hours the delay comes up again. It is possible, that i get an old frame (one second ago) with switch_core_session_read_frame ? Is there a solution (for example an freeswitch core api call) to avoid the delay ? Greetings Achim Stamm -- Achim Stamm, Dipl.-Inform. (FH) Lyncker & Theis GmbH Wilhelmstr. 16 65185 Wiesbaden Germany Fon +49 611/9006951 Fax +49 611/9406125 Handelsregister: HRB 23156 Amtsgericht Wiesbaden Steuernummer: 04323897052 USt-IdNr.: DE255806399 Geschäftsführer: Filip Lyncker, Armin Theis

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[Freeswitch-dev] Everyone come join IRC!
by Michael Collins 26 Aug '10

26 Aug '10

Hey all, We are having a big IRC day today so we'd love to have everyone join and help answer questions. Please join #freeswitch on irc.freenode.net. Thanks! -MC

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[Freeswitch-dev] Unsubscribe
by Robert Clayton 25 Aug '10

25 Aug '10

When moving both my Freeswitch dev and user accounts it appears that one must log in to unsubscribe as both the unsubscribe and remind (password) buttons do not work. The problem being if a person does not remember their password they would not be able to unsubscribe. Which is the problem with my dev account.

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[Freeswitch-dev] How is MOH implemented in Freeswitch
by sushil verma 24 Aug '10

24 Aug '10

Hi, It seems Freeswitch implements MOH (music on hold) feature using sofia module. Reference: http://wiki.freeswitch.org/wiki/Music_on_Hold Can someone explain (or point me to some design/feature document etc of freeswitch) how it has been implemented? Regards, Sushil Kumar

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[Freeswitch-dev] FreeSWITCH Conference Call Starting Shortly!
by Michael Collins 18 Aug '10

18 Aug '10

C'mon down! http://wiki.freeswitch.org/wiki/FS_weekly_2010_08_18 Brian K West will be by to talk about NAT and VoIP with FreeSWITCH. -MC

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[Freeswitch-dev] Socket - bridge answered event
by Ali Roumie 16 Aug '10

16 Aug '10

Hello All, This is my first post to this list and many thanks to all contributors to this state of art project. I'm using outbound socket and everything is going wonderful with me except one thing so far. My logic is simple, collect a PIN from Leg A and once the PIN is authorized, I bridge the call with a SIP provider. Everything is great, call got bridged successfully (and I was supper exited when it worked) but my problems is FS is not sending my socket an event when the Leg B is answered. I get an event only when Leg B is hangup. I must mention, I set filters on the socket to avoid lots of the many generated events by FS Here is my command list on the socket. connect myevents filter Event-Name CHANNEL_ANSWER filter Event-Name CHANNEL_CREATE filter Event-Name CHANNEL_EXECUTE_COMPLETE filter Event-Name CHANNEL_BRIDGE filter Event-Name CHANNEL_UNBRIDGE filter Event-Name CHANNEL_HANGUP sendmsg call-command: execute execute-app-name: answer sendmsg call-command: execute execute-app-name: sleep execute-app-arg:1000 event-lock:true sendmsg call-command: execute execute-app-name: play_and_get_digits execute-app-arg:2 5 3 5000 # ivr/ivr-please_enter_pin_followed_by_pound.wav ivr/ivr-please_reenter_your_pin.wav chanDTMF ^\d{5}$ event-lock:true sendmsg call-command: execute execute-app-name: bridge execute-app-arg:[sip_auth_username=XXXX,sip_auth_password=*******]sofia/internal/123456789@PROVIDER.COM event-lock:true Many Thanks, Ali R.

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[Freeswitch-dev] New Hardware for Freeswitch
by Ari Siitonen 16 Aug '10

16 Aug '10

Hello all! I would like to announce a new hardware development project: - ISDN Pri interface card 1-8 channels with/without QUICC for ISDN stack + dsp - 1-8 POTS line card - ISDN Bri line card network/user side Our company has manufactured such cards and IVR-server back in last century, and we would like to make the "new generation" cards compatible with freeswitch. So, to start Hardware redevelopment in right direction, any pointers and suggestions on driver design would be greatly appreciated. And if the existing cards lack some features, we will try to incorporate them (hw conference bridging?) Also, pilot users will be given free cards to test and evaluate. Best Regards, Ari Siitonen Voice Stream Oy Finland

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[Freeswitch-dev] Sip RFC2543 and FreeSwitch
by Tech Micron 16 Aug '10

16 Aug '10

Hi everyone, I have a SIP device that is working with RFC 2543. I am trying to send call to FreeSwitch, but none of my calls are going through. As I activated the debug on SOFIA, it returns an error as below. Can anyone help me to solve the problem? any hints? recv 660 bytes from udp/[66.220.15.230]:5060 at 18:45:21.239029: ------------------------------------------------------------------------ INVITE sip:1000@66.220.15.234 SIP/2.0 v: SIP/2.0/UDP 66.220.15.230 To: sip:1000@66.220.15.234 <sip:1000@66.220.15.234> From: john odu <sip:26679778@66.220.15.230>;tag=4c5c58410003ac2076f6302c11fc3b7b i: 41585c4c2ba3030035e697f0a586dbaa(a)66.220.15.230 CSeq: 1 INVITE m: john odu <sip:26679778@66.220.15.230> Supported: com.lilo.conferencing, com.lilo.mux User-Agent: liloSipMCU/3.1.6.200801281402 c: application/sdp l: 212 v=0 o=mcu 0 0 IN IP4 66.220.15.230 s=sip:26679778@66.220.15.230 c=IN IP4 66.220.15.230 t=0 0 m=audio 2006 RTP/AVP 0 111 a=ptime:20 a=rtpmap:0 PCMU/8000 a=rtpmap:111 telephone-event/8000 a=xlilossrc:8 ------------------------------------------------------------------------ tport_deliver(0x97a1fb0): msg 0xb762bef8 (660 bytes) from udp/66.220.15.230:5060/sip next=(nil) nta: received INVITE sip:1000@66.220.15.234 SIP/2.0 (CSeq 1) nta: INVITE has bad To header nta: INVITE (1) is Bad To Header tport_tsend(0x97a1fb0) tpn = UDP/66.220.15.230:5060 tport_resolve addrinfo = 66.220.15.230:5060 tport_by_addrinfo(0x97a1fb0): not found by name UDP/66.220.15.230:5060 tport_vsend(0x97a1fb0): 239 bytes of 239 to udp/66.220.15.230:5060 tport_vsend returned 239 send 239 bytes to udp/[66.220.15.230]:5060 at 18:45:21.239369: ------------------------------------------------------------------------ SIP/2.0 400 Bad To Header Via: SIP/2.0/UDP 66.220.15.230 From: john odu <sip:26679778@66.220.15.230>;tag=4c5c58410003ac2076f6302c11fc3b7b Call-ID: 41585c4c2ba3030035e697f0a586dbaa(a)66.220.15.230 CSeq: 1 INVITE Content-Length: 0

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