January 2014 - FreeSWITCH-users - FreeSWITCH Mailing Lists

conference user
by Joli Martinez 24 Jan '14

24 Jan '14

Hello, I need to know the userid number of a person joining a conference. Is there a way to get that userid number when a person joins the conference and save it as a variable in Lua? Thanks

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Join us for the Friday FreeSWITCH Bug Hunt!
by Ken Rice 24 Jan '14

24 Jan '14

Hey FreeSWITCHers, Join us today 2PM EST (11AM PST) for the Friday FreeSWITCH Bug Hunt! Where? sip:888@conference.freeswitch.org or see http://fs0.us/call888 What? FreeSWITCH Bug Hunt, Jira Reviews, and General FS Support! Duration? As Long as it takes today! Help us help you, Join the Bug Hunt Today! -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH

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4 second delay in call.
by Andrew 24 Jan '14

24 Jan '14

I know most of the people don't like windows but I am looking for some help. I am building a switch and the exe is done in .net. I have everything in threads. I found that I will take in a call then do a bridge all in less than 1 second. When doing a wire shark trace it looks like the bridge call take almost 4-6 seconds to send a request. I am close to giving up. I did a study and traced every step of my code and its clear its not .net due to the call will come in and out in less than a sec. Again all of this from incoming to bridge happens in less than 1 second but wire shark shows almost 4 seconds from incoming to outgoing call. string Route = "{bypass_media=" + MediaByPass + ",hangup_after_bridge=true,hangup_complete_with_xml=true,continue_on_fail=tr ue,execute_on_answer=sched_hangup +" + SecMaxTalkTime + " alloted_timeout, ignore_early_media=true, origination_caller_id_number=" + CallerIDNumber + ", origination_caller_id_name=" + CallerIDName + ",originate_timeout=" + RingTimeOUT + ",call_timeout=" + RingTimeOUT + ",progress_timeout=" + PDD + ",fail_on_single_reject=USER_BUSY NO_ANSWER NO_USER_RESPONSE ORIGINATOR_CANCEL} " + DialPath; ManagedSession leg_A = new ManagedSession(UUID_LegA); // leg_A.begin_allow_threads(); leg_A.SetVariable("bypass_media", MediaByPass.ToString()); leg_A.Execute("bridge", Route);

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How to refuse T.38 requests
by gegetel 24 Jan '14

24 Jan '14

Hello, My locale UAs do not know anything about T.38 protocol and I want to send a fax to some PSTN subscriber via OVH, my french VoIP provider, in an RTP/PCMA session. This session begins normally but, as soon as OVH's gateway recognizes fax tones from remote and/or locale side, an undesirable UPDATE with SDP for a T.38 session is sent to my side. The problem is that, immediately, FreeSwitch says "200 OK" instead of something like "488 Not acceptable here" and this is the beginning of a dialogue of the deaf which only takes end when remote fax-machine hangs up. I have done several things that I found in the WIKI: In my dialplan added these lines: <action application="set" data="sip_ignore_reinvites=true"/> <action application="set" data="fax_enable_t38=false"/> <action application="set" data="fax_enable_t38_request=false"/> In ~/conf/vars.xml: <X-PRE-PROCESS cmd="set" data="refuse_t38=true"/> In ~/conf/sip_profiles/external/sip.ovh.fr.xml: <param name="t38-passthru" value="false"/> But without better result. This is an example of the UPDATE request: UPDATE sip:gw+sip.ovh.fr@92.146.52.82:5080;transport=udp;gw=sip.ovh.fr SIP/2.0 Call-ID: 9686bdc8-e997-1231-27aa-001fc61584f4 Contact: <sip:10.7.1.60:5060> Content-Type: application/sdp Cseq: 29265730 UPDATE From: <sip:0556ZZZZZZ@sip.ovh.fr>;tag=00-08184-01c44a83-0e1305300 Max-Forwards: 30 Record-Route: <sip:91.121.129.20:5060;lr> To: \"0033972XXXXXX\" <sip:0033972XXXXXX@sip.ovh.fr>;tag=75761tHD7541g Via: SIP/2.0/UDP 91.121.129.20:5060;branch=z9hG4bK-DIVG-481b17bf-03a84e77 User-Agent: Cirpack/v4.56 (gw_sip) Content-Length: 299 v=0 o=cp10 138814832168 138814832170 IN IP4 10.7.1.132 s=SIP Call c=IN IP4 91.121.128.144 t=0 0 m=image 35996 udptl t38 a=sendrecv a=T38FaxVersion:0 a=T38MaxBitRate:9600 a=T38FaxRateManagement:TransferredTCF a=T38FaxMaxBuffer:1000 a=T38FaxMaxDatagram:200 a=T38FaxUdpEC:t38UDPRedundancy And that is the OK answer: SIP/2.0 200 OK Via: SIP/2.0/UDP 91.121.129.20:5060;branch=z9hG4bK-DIVG-481b17bf-03a84e77 Record-Route: <sip:91.121.129.20:5060;lr> From: <sip:0556ZZZZZZ@sip.ovh.fr>;tag=00-08184-01c44a83-0e1305300 To: \"0033972XXXXXX\" <sip:0033972XXXXXX@sip.ovh.fr>;tag=75761tHD7541g Call-ID: 9686bdc8-e997-1231-27aa-001fc61584f4 Cseq: 29265730 UPDATE Contact: <sip:gw+sip.ovh.fr@92.146.52.82:5080>;transport=udp;gw=sip.ovh.fr User-Agent: FreeSWITCH-mod_sofia/1.5.8b+git~20131213T181356Z~87751f9eaf~64bit Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Content-Type: application/sdp Content-Disposition: session Content-Length: 201 v=0 o=FreeSWITCH 1388128642 1388128643 IN IP4 92.146.52.82 s=FreeSWITCH CH c=IN IP4 92.146.52.82 t=0 0 m=audio 19678 RTP/AVP 8 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 Any help much appreciated, thanks in advance...

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lua script after transfer to conference
by Joli Martinez 24 Jan '14

24 Jan '14

I have a Lua script that get the conference_member_id and then passes it to a DB. FS wont give me a member_id until I transfer the call to the conference and once I transfer the call to the conference the script stops execution until the user hangs up. How can I get the lua script to continue execution even after I transfer a call to the conference? I have session:execute("conference", 3001); userId = session:getVariable("conference_member_id") Thanks

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evaluate
by mbo 23 Jan '14

23 Jan '14

Hello, in a bridge I can evaluate a sofia contact like <action application="bridge" data="${sofia_contact(internal/100(a)sys.local)}^222" /> that works fine. If I try to use the same approach in a originate like: originate ${sofia_contact(user/1000)} &park I can see: 2014-01-23 20:14:18.216587 [DEBUG] switch_ivr_originate.c:2050 Parsing global variables 2014-01-23 20:14:18.216587 [ERR] switch_core_session.c:496 Could not locate channel type ${sofia_contact(user 2014-01-23 20:14:18.216587 [NOTICE] switch_ivr_originate.c:2651 Cannot create outgoing channel of type [${sofia_contact(user] cause: [CHAN_NOT_IMPLEMENTED] 2014-01-23 20:14:18.216587 [DEBUG] switch_ivr_originate.c:3622 Originate Resulted in Error Cause: 66 [CHAN_NOT_IMPLEMENTED] Is there a way to resolve/evaluate sofia_contact also in an originate? Thanks Markus

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Change default extensions
by peetee23＠hushmail.com 23 Jan '14

23 Jan '14

Dear All, could you please explain me what I have to change in dialplan to use SIP extensions 1-19 instead of default 1000-1019? Above of that what is essential in dialplan to have working internal SIP calls and maybe SMS/SIMPLE messages only? Thanks a lot for any feedback! Kind regards, PT

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freeswitch in sandbox
by Joli Martinez 23 Jan '14

23 Jan '14

Hello, I have a python script located in /usr/local/freeswitch/scripts and a lua script calls the python script. Scripts works when calling lua <scriptname>, but when invoking it from freeswitch I get "file not found" on the python script. permissions are set to 755 and owner is freeswitch:daemon. both the Lua and the python script are in the same scripts directory. The python script makes a mysql call and parses some data and return it back to Lua. I remember having this problem before and I believe that freeswitch was in a sandbox enviroment, but dont remember what I changed. Thanks,

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Caller ID unknown with Sangoma A200
by Peter Hartmann 23 Jan '14

23 Jan '14

Hi, Is there anything special I need to do to enable caller id on a Sanagoma A200 analog card? Everything is working with this card except for incoming caller id. I have <param name="enable-callerid" value="true"/> in the fxo section of my conf/autoload/freetdm.conf.xml. By the way, I have also tried underscore....I've seen it both ways on Sangoma's site and the wiki. I put a snippet of the log here: http://pastebin.com/nAsr0wQj There's plenty mention of GET_CALLERID but when it comes time to refer to the caller it always comes up 'unknown' like so. [INFO] mod_dialplan_xml.c:558 Processing unknown <unknown>->212XXXXXXX in context public By the way, this is using Version 1.5.8b git a26fdca, and this is in the NYC area. I'm assuming we get the fsk between rings one and two. I though it's curious that I haven't had to specify what type of caller id I'm using in wanpipe via wancfg_fs, wancfg or in FS. I did see references to bell202 and other types in the wanpipe api sources. Does FS tell wanpipe to use Bellcore by default? Thanks much in advance, Peter Hartmann Hartmann Computer Consulting http://blog.hartmanncomputer.com

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rtp-timer-name none packet loss handling
by Jozsef Vass 23 Jan '14

23 Jan '14

We are using freeswitch to transcode between opus and g711. We have rtp-timer-name set to none in sip_profiles. If we experience packet loss on the inbound leg (opus), freeswitch will take no action. When the next opus packet arrives, freeswitch sends a pcmu rtp packet with continuous sequence number and timestamp to the outbound leg. That is, there is no way to tell from the outbound leg that there was a packet loss on the inbound leg. I would prefer the following behavior: 1. When there is a packet loss on the inbound leg, increase the sequence number and timestamp on the outbound leg. 2. When there is a packet loss on the inbound leg, interpolate the missing packet using opus plc. This only works for a few consecutive lost packets and not for burst loss. Is there any config parameters to force either of the above behavior? Jozsef

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