January 2011 - freeswitch-dev@lists.freeswitch.org

[Freeswitch-dev] RTCP patch to send source reports in the RECV_RTCP_MESSAGE event
by Guillaume Binet 21 May '11

21 May '11

Hi all, Please find attached a patch that parses and add RTCP source reports and added the info to the corresponding the switch event. It adds the following headers, numbered by source : Event arrived : RECV_RTCP_MESSAGE Event Header : Event-Name -> RECV_RTCP_MESSAGE Event Header : Core-UUID -> 649cbe4e-7bb8-4edd-a0f5-2662246966f8 Event Header : FreeSWITCH-Hostname -> sal Event Header : FreeSWITCH-IPv4 -> 192.168.0.89 Event Header : FreeSWITCH-IPv6 -> ::1 Event Header : Event-Date-Local -> 2010-06-29 12:11:00 Event Header : Event-Date-GMT -> Tue, 29 Jun 2010 10:11:00 GMT Event Header : Event-Date-Timestamp -> 1277806260737911 Event Header : Event-Calling-File -> mod_sofia.c Event Header : Event-Calling-Function -> sofia_read_frame Event Header : Event-Calling-Line-Number -> 907 Event Header : Unique-ID -> 4fa7d00b-64f2-47ff-b2db-266943ee5fe2 Event Header : SSRC -> 402076d6 Event Header : NTP-Most-Significant-Word -> 3486795060 Event Header : NTP-Least-Significant-Word -> 3111085330 Event Header : RTP-Timestamp -> 40160 Event Header : Sender-Packet-Count -> 246 Event Header : Octect-Packet-Count -> 39360 Event Header : Last-RTP-Timestamp -> 39840 Event Header : RTP-Rate -> 8000 Event Header : Capture-Time -> 1277806260738059 --> HERE Event Header : Source0-SSRC -> 18320497 Event Header : Source0-Fraction -> 0 Event Header : Source0-Lost -> 0 Event Header : Source0-Highest-Sequence-Number-Received -> 40032 Event Header : Source0-Jitter -> 5 Event Header : Source0-LSR -> 0 Event Header : Source0-DLSR -> 0 Guillaume BINET. Guillaume Binet Mobile: +32 487 57 35 81 Fax: +32 2 223 01 15 Ch. de la Hulpe 181 Terhulpsesteenweg Bruxelles B-1170 Brussel www.mondialtelecom.be To discover the innovation of Mondial Telecom, becherry.be

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[Freeswitch-dev] Bridge to PSTN Number Not Using Call Timeout Value
by Jerry Richards 15 Feb '11

15 Feb '11

Hello All, When I set the call_timeout variable (to 12 seconds) before a bridge to an outbound PSTN call, Freeswitch always rings for 24 seconds before going to voicemail. If I call an internal extension, then it uses the call_timeout variable. Do you know why it would do this? Here is the line that sets the call_timeout variable: switch_channel_set_variable(channel, "call_timeout", "6"); Here is the PSTN bridge log: EXECUTE sofia/internal/2003@192.168.72.79:5060 bridge({presence_id=2001(a)192.168.72.79}[lcr_carrier=Carrier / Location 1/INTERNAL PRI TRUNK,lcr_rate=0.00200,origination_caller_id_number=2003]openzap/smg_prid/a/15248142341@g1) Thanks, Jerry

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[Freeswitch-dev] SIP INFO carrying MSML
by Owen Williams 10 Feb '11

10 Feb '11

Hi all, I'm looking into the possibility of getting basic msml support into freeswitch. We have a few applications hosted on a remote SIP Application server that already use MSML against another media server. The application acts as a back2back between a caller and the media server and sends info messages (inside an invite dialog) to the media server with the INFO sdp containing MSML (e.g. play file, join conf, etc). I'm a bit new to freeswitch but I'm guessing mod_sofia would need to modified to send info events to a new module or to the ESL, and mod_sofia also needs to be capable of receiving events to send INFOs back to the remote B2B. There was a bounty completed about 18 months ago (Generate and be notified of SIP INFO messages) however this may have been only for out of dialog INFOs. http://wiki.freeswitch.org/wiki/Bounty#Generate_and_be_notified_of_SIP_INFO… Any information as to whether the above bounty made it into trunk and how it can be enabled would be appreciated. I'd like to have a play with writing a MSML processor through ESL as I've far more experience with ESL than writing c modules. Regards, Owen

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[Freeswitch-dev] offering two "m=" lines from FreeSWITCH
by Ujjwal SIngh 09 Feb '11

09 Feb '11

Hi, I am new to FreeSWITCH, I was just wondering that can we initiate Best Effort SRTP offer from FreeSWITCH in the same way as Polycom phones do, i.e sending two "m=" lines for the same media, the first line having a SAVP profile, while the second one having the normal AVP profile. Kindly please let me know how can i do this. Thanks in advance. Regards, Ujjwal

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[Freeswitch-dev] switch_xml_config.c
by Gilles Depatie 06 Feb '11

06 Feb '11

Hi I'm running into what appears to be a questionable range interpretation issue in switch_xml_config_parse_event function. When enforce_min and enforce_max are TRUE should the value not be inclusive of MIN and MAX? Say a value range of min = '0' and max = '1000', should values of 0 and 1000 not be acceptable? The current statement: if ((int_options->enforce_min && !(intval > int_options->min)) || (int_options->enforce_max && !(intval < int_options->max))) excludes those values presently The error message seems to imply that the range is inclusive of the min and max values I've had a look at the latest git version or the file and in this respect nothing has changed from the version I am currently using GILLES Gilles Depatie, Senior S/W Designer, MapleWorks Technology O: 819-776-6066 x353 | gilles.depatie(a)mapleworks.com

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[Freeswitch-dev] Developer Dinner Coming Up - Help Feed The FreeSWITCH Team!
by Michael Collins 01 Feb '11

01 Feb '11

Well, it's that time again. The FreeSWITCH developers are gathering at a secret location in Wisconsin next week and the community is invited to participate in this year's "buy the devs dinner" event. How to do it: Go to freeswitch.org and click on the Paypal link. Toss a few bucks in the hat and send along your bon apetit message. If for some reason you cannot use Paypal but still wish to participate then by all means call Brian West on his SIP phone: 2000(a)bkw.org. (Hint: you can dial 9191 if you have a default FreeSWITCH config.) Brian is well-prepared to take your money. :) To those who have donated recently we say, "Thank you!" If you wish to donate again specifically for this occasion you are more than welcome to do so. In any case, thank you in advance for your generosity. You are great community members and the FreeSWITCH team is absolutely glad to have you with us! -Michael Collins

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[Freeswitch-dev] freeswitch.com is returned to us!
by Anthony Minessale 31 Jan '11

31 Jan '11

http://www.freeswitch.com/ Thank you everyone who helped with this! -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900

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[Freeswitch-dev] Freeswitch on ARM, Memory alignment issues
by M D 28 Jan '11

28 Jan '11

Summary: Has anyone looked at/actively looking at the 32-bit boundary memory alignment requirement for the ARM architecture? Detail: I am running freeswitch on an ARM platform (dockstar based on Marvell Kirkwood). I have compiled and tested both freeswitch-1.0.6 and freeswitch-head(1,2), and found that stun lookups were failing with a "Timeout" error. Debugging the issue points to memory alignment issues specific to the ARM architecture. Basically, due to the fact that the uint8_t array "buf", is not aligned at a 32-bit memory boundary, the stun lookup packet sent to the stun server was malformed. Web searches confirm that a symptom of misaligned access on ARM is an apparent memory corruption. This can be fixed by forcing alignment of specific variables to 32-bit boundary. As an example, in the switch_stun_lookup function of file src/switch_stun.c , we currently have { switch_sockaddr_t *local_addr = NULL, *remote_addr = NULL, *from_addr = NULL; switch_socket_t *sock = NULL; uint8_t buf[260] = { 0 }; .... } If I change it to { switch_sockaddr_t *local_addr = NULL, *remote_addr = NULL, *from_addr = NULL; switch_socket_t *sock = NULL; uint8_t buf[260] __attribute__ (aligned(4)) = { 0 }; .... } or { switch_sockaddr_t *local_addr = NULL, *remote_addr = NULL, *from_addr = NULL; switch_socket_t *sock = NULL; uint8_t *buf; buf = (uint8_t*) malloc(sizeof(uint8_t)*260); .... } the stun lookup packet sent is correct.freeswitch is also able to parse the recvd packet and correctly determines the external IP. Random crashes coupled with looking at the code in switch_rtp suggests that the memory alignment issue may not be limited to this function or even this file. Has anyone looked at ARM memory alignment issues for freeswitch in general? cheers, -m (1) Current head leads to the situation where call gets connected but there is no-audio. I have not had the chance to debug this, but suspect this could be due to the "eat rtp .... " patch combined with ARM memory alignment issues. (2) The tested (audio working, some issues) git version is commit 715d250e171a94736b19019ac742f739899ad997 Author: Mathieu Parent <math.parent(a)gmail.com> Date: Wed Dec 15 21:29:52 2010 +0100

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[Freeswitch-dev] fsctl pause
by Bernhard Suttner 27 Jan '11

27 Jan '11

Hi, if I use "fsctl pause" freeswitch should not handle any new call but a INVITE will still be send answered from FreeSWITCH with 407 Proxy Authentication Required. This message go back to the peer. If the peer then send the INVITE with credentials (with authentication) FreeSWITCH sends bacck the 503 "No more sessions" because of the "pause state". Shouldn´t freeswitch respond with 503 for the first INVITE and not only for the second INVITE with authentication? Best regards, Bernhard

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[Freeswitch-dev] *FreeSWITCH Conf Call Now! Hear About New Vestec Speech Rec Engine!
by Michael Collins 26 Jan '11

26 Jan '11

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