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

[Freeswitch-dev] mod_socket
by Ali R. 05 Jul '11

05 Jul '11

Hi everyone I think my issue could be fixed with mod_xml_curl but I'm using FS purely through sockets (inbound mode) When I get the park event, I execute some biz logic. Leg's A source IP plays a role in this logic so based on the value of the IP (a big pool of allowed IP addresses is generated dynamically and changes very fast so I have a logic to query this pool) all good so far. However, if the incoming channel’s source IP is not allowed I would still want to answer and continue on but I must challenge it with a user name and password that I have a logic to retrieve. Any thoughts on how should I go about doing the username/password challenge through the socket without starting a new listening socket for mod_xml_curl requests . I really appreciate any thoughts on this issue Many Thanks,

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[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] Implementing Forward-{All, Busy, NoAnswer}, ring policies and DND in FS
by Mathieu Parent 14 May '11

14 May '11

Hello, *Context*: --------- I want to implement the following in mod_skinny (FS-3048 and FS-3047) in a way shared with other endpoints modules: - call forwarding[cfw], - DND and ring policies *Call forwarding*: --------- There are three main kinds of forwarding: - forward all (immediately) - forward when line is busy - forward when no answer We should also store for each the state of the forwarding (enabled or not). I think a reasonable default here would be: - to forward "no answer" to voicemail (and enabled) - to forward busy to voicemail (but disabled) - to forward all to "" (and disabled) We also should define what a busy line is, especially on shared lines (perhaps define a busy threshold, defaulting to 1). Call forwarding should be implemented in dialplan, but endpoint modules should know when a call is forwarded (the bridge event is probably sufficient). *Ring policies and DND*: --------- A do-not-disturb flag can be set on an extension so that the phone doesn't ring (but the user still knows he/she is called). DND is a specific case of ring policies. I propose the following default ring policies: - on idle: ring, blink (toggled to blink with DND button) - on busy: blink only (also available: ring once) This should be implemented per endpoint module. *Proposal*: --------- I propose to use hashes to store the various data: <action application="hash" data="insert/${realm}/${extension}/${data}"/> Where ${realm} is one of: - ${domain_name}-forward-{all,busy,noanswer}-destination: a destination number - ${domain_name}-forward-{all,busy,noanswer}-status: true/false - ${domain_name}-busy-threshold: [0-9]+ - ${domain_name}-ring-on-idle: true/false/once - ${domain_name}-blink-on-idle: true/false - ${domain_name}-ring-on-busy: true/false/once - ${domain_name}-blink-on-busy: true/false The dialplan should be enhanced to manage forwarding. The endpoint modules should be enhanced to manage forwarding (notify that the call has been forwarded) and ring policies. Does this makes sense? How can we improve it? *Next steps*: --------- - proposal discussion - forward-* dialplan implementation (I need help here, I am not an dialplan expert) - mod_skinny forward-* implementation FS-3048 - mod_skinny ring policies implementation FS-3047 - other endpoints implementation Regards Mathieu [cfw]: http://en.wikipedia.org/wiki/Call_forwarding [dnd]: http://en.wikipedia.org/wiki/Do_Not_Disturb_%28telecommunications%29

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[Freeswitch-dev] using freeswich as for SIP signalling and programming an external HW.
by Narendra Sirugudi 12 May '11

12 May '11

Hi All, I want to use freeswich for SIP signalling and program an external HW for RTP processing. Does freeswitch provide any mechanism/hooks to program an external HW for RTP processing ? thanks, --kumar

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[Freeswitch-dev] NEW skypopen installer and easy Skype client download, Skype calls on FreeSWITCH in one minute
by Giovanni Maruzzelli 30 Apr '11

30 Apr '11

Dear FreeSWITCHers, after a fair amount of effort, I ended up with a new way to install and use mod_skypopen on Linux. No more looking around the internet for the lost 2.0.0.72 Skype client for ALSA. First, we can use the readily available Skype client for OSS. (added benefit: no more need for ALSA driver. I wrote skypopen.ko OSS driver, that's very easy to compile and install, and do not need to mess with the operating system installation.) Second, I wrote an installer that automatically do all the tedious work for you: download and install the skype client, create the config directory for Skype clients, create the config file for mod_skypopen, create the script that launches the Skype clients. I hope those improvements will lower the barriers for Skype calls on FreeSWITCH. Actually is ludicrously simple now, and after you compile FreeSWITCH, mod_skypopen and the skypopen.ko OSS driver it will take like less than one minute to have a complete installation of mod_skypopen ready to make and receive calls. All automatic, no more need to fiddle around with the Skype client download, configurations, authorization, etc. Is all well tested, but maybe there are still some bugs, and maybe the docs are not clear/easy enough. Please have a look at the new and improved wiki page and let me know what do you think about (and maybe test the procedures). http://wiki.freeswitch.org/wiki/Mod_skypopen_Skype_Endpoint_and_Trunk http://wiki.freeswitch.org/wiki/Mod_skypopen_Skype_Endpoint_and_Trunk#Linux http://wiki.freeswitch.org/wiki/Mod_skypopen_Skype_Endpoint_and_Trunk#Inter… You must update to the latest git to have all the goodies. Thank you all for your support, -giovanni -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618

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[Freeswitch-dev] Reviewers needed: FreeSWITCH Cookbook
by Michael Collins 29 Apr '11

29 Apr '11

Hello, As you know we are working on a FreeSWITCH Cookbook for Packt Publishing. We are in need of 3 or 4 capable reviewers. Please contact me off list if you want to help and meet these qualifications: * Read and write English fluently * Have at least one instance of FreeSWITCH (or a sandbox) where you can test each recipe * Have time to read and try all the recipes in the book * Can offer useful feedback and proofreading * Preferably have a later version of MS Word for editing If you meet the above qualifications and would like to help out then please contact me off list for more information. Thanks, MC

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[Freeswitch-dev] mod_voicemail new API functions
by João Mesquita 29 Apr '11

29 Apr '11

Guys, I see that Moc has added new API functions to set preferences to the mailbox. Those work great and are awesome, but I think there is one small feature missing. Shouldn't we be able to remove the greetings instead of just setting new ones so that mod_voicemail restores it's original configuration? I don't have any problem hacking in the feature, I am just making sure I am not going to do something that is already in there and that I have overlooked before. Thanks! João Mesquita

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[Freeswitch-dev] 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

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[Freeswitch-dev] 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.

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[Freeswitch-dev] 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

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