September 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

limit_usage change in recent versions?
by Yehavi Bourvine 19 Sep '12

19 Sep '12

Hello, I am using limit_usage in the following format: <action application="bridge" data="${cond(${limit_usage(db, max_calls ${destination_number})} > ${MaxLimit} ? error/user_busy : ${destination_mumber})}" /> It worked at older versions (my local reference is 1.0.6), but it doesn't work in newer ones. It always returns zero. All I want is to know whether the extension already has an active call to decide what to do in case it is busy. What is the correct way of calling it? Thanks, __Yehavi:

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t38 issue
by royj 19 Sep '12

19 Sep '12

Thank you for all your assistance Yesterday FreeSWITCH have been updated to Version 1.3.0+git~20120918T210019Z~d80e91d248 (1.3.0; git at commit d80e91d248 on Tue, 18 Sep 2012 21:00:19 Z) and no changes. Maybe there is some specific settings, that I miss? Does anybody has been solving this mission? On 9/11/12 13:36 AM, "Ken Rice" <krice(a)freeswitch.org> wrote: > > There have been updates to t.38 since 1.2.0-rc2... You should either be on > head or v1.2.stable branch > > > > > On 9/11/12 10:35 AM, "Michael Collins" <msc(a)freeswitch.org> wrote: > > > I think the guys have made some improvements since 1.2.0-rc2. Any chance you > > can get at least up to 1.2.1-stable and retest? > > -MC > > > > On Tue, Sep 11, 2012 at 1:48 AM, royj <royj(a)yandex.ru> wrote: > >> > >> Hi all > >> > >> There is some strange issue with t38-passthru (FreeSWITCH Version 1.2.0-rc2). > >> Task - t38-passthru > >> spa112(ip 1.1.1.1) <--(profile local)--> FreeSWITCH(ip 2.2.2.2)<--(profile > >> extrernal)--> Mediant 2000(ip 3.3.3.3); fax from spa112 to Mediant 2000. > >> In both profiles: > >> <param name="t38-passthru" value="true"/> > >> <param name="inbound-proxy-media" value="true"/> > >> <param name="inbound-late-negotiation" value="true"/> > >> > >> Here is the sip-trace of the call with fax - > >> http://pastebin.freeswitch.org/19866 > >> > >> It is seen that t38 re-INVITE from spa112 (profile local) passed on and back > >> with "200 OK", but t38 from Mediant 2000 (profile external) is not (in > >> response to re-INVITE t38 (SDP, m=image) - "200 OK" with SDP m=audio). After > >> agreement the SDP, the data (v21-preamble) is not passed by freeswitch > >> between spa112 and Mediant 2000, and one of the party ends conversation with > >> message "no-signal" and then "by" . > >> > >> Wiki says ( > >> http://wiki.freeswitch.org/wiki/Proxy_Media#Why_you_almost_certainly_don.27… > >> want_to_use_it ) that module mod_spandsp is required (if I understand > >> correctly), but "load mod_spandsp" does not change anything. > >> > >> I would appreciate your help > >> > >> -- > >> Regards > >> royj > >> > >> _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting(a)freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> FreeSWITCH-powered IP PBX: The CudaTel Communication Server > >> http://www.cudatel.com > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> http://www.cluecon.com > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users(a)lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > > > > > > -- > Ken > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > irc.freenode.net #freeswitch > >

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mod_conference wrong codec for auto_outcall b-leg
by Oliver Saggau | Sysvision GmbH 19 Sep '12

19 Sep '12

Hi everyone, I got a strange problem with mod_conference and auto_outcall taking the wrong codec for b-leg. For what I can see from the log everything seems fine. The a-leg calls with the following SDP: 2012-09-18 08:57:47.410023 [DEBUG] sofia.c:6293 Remote SDP: v=0 o=- 3556961565 3556961565 IN IP4 10.20.30.125 s=pjmedia t=0 0 m=audio 4000 RTP/AVP 99 101 c=IN IP4 10.20.30.125 a=rtpmap:99 SILK/24000 a=fmtp:99 useinbandfec=0 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=rtcp:4001 IN IP4 10.20.30.125 And the SILK/24000 codec is accepted according to the log file: 2012-09-18 08:57:47.650018 [DEBUG] sofia_glue.c:3077 Set Codec sofia/internal/1000004(a)10.20.30.240 SILK/24000 20 ms 480 samples 40000 bits 2012-09-18 08:57:47.650018 [DEBUG] switch_core_codec.c:111 sofia/internal/1000004(a)10.20.30.240 Original read codec set to SILK:120 2012-09-18 08:57:47.650018 [DEBUG] mod_sofia.c:836 Local SDP sofia/internal/1000004(a)10.20.30.240: v=0 o=FreeSWITCH 1347951445 1347951446 IN IP4 10.20.30.240 s=FreeSWITCH c=IN IP4 10.20.30.240 t=0 0 m=audio 21622 RTP/AVP 99 101 a=rtpmap:99 SILK/24000 a=fmtp:99 useinbandfec=0; usedtx=0; maxaveragebitrate=40000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv The conference get's created and activates the read/write codecs for the first member: 2012-09-18 08:57:47.990021 [DEBUG] mod_conference.c:7092 Raw Codec Activation Success L16@24000hz 1 channel 20ms 2012-09-18 08:57:47.990021 [DEBUG] mod_conference.c:7137 Raw Codec Activation Success L16@48000hz 1 channel 20ms After that he's trying to do the auto_outcall and that's where things get weird: EXECUTE sofia/internal/1000004(a)freeswitch.aws.hosts.corpex.de conference_set_auto_outcall(sofia/internal/sip:1000003@10.20.30.108:63741;transport=UDP;ob) 2012-09-18 08:57:47.990021 [NOTICE] switch_channel.c:951 New Channel sofia/internal/sip:1000003@10.20.30.108:63741 [0236d005-1ff0-4f13-9370-bf7736087dd0] 2012-09-18 08:57:47.990021 [DEBUG] sofia_glue.c:2637 Local SDP: v=0 o=FreeSWITCH 1347952749 1347952750 IN IP4 10.20.30.240 s=FreeSWITCH c=IN IP4 10.20.30.240 t=0 0 m=audio 20318 RTP/AVP 98 0 8 101 13 a=rtpmap:98 SILK/8000 a=fmtp:98 useinbandfec=1; usedtx=0 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv 2012-09-18 08:57:52.590016 [DEBUG] mod_conference.c:7092 Raw Codec Activation Success L16@8000hz 1 channel 20ms 2012-09-18 08:57:52.590016 [DEBUG] mod_conference.c:7137 Raw Codec Activation Success L16@48000hz 1 channel 20ms Why is FS only offering SILK/8000 for b-leg? My outbound_codec_prefs is configured like this: outbound_codec_prefs=silk@24000,silk@16000,silk@8000,PCMU,PCMA,GSM Thanks for your help, Oliver

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You don't need to save a greeting to override your existing one in Voicemail
by Emrah 19 Sep '12

19 Sep '12

It used to be the case with Asterisk as well, now it has changed. Basically, if you have a greeting in place and want to record a new one… And you suddenly cough in the middle of your recording… Than get interrupted by your boss right after you've pressed #… You just hang up the phone and think that your existing greeting hasn't been altered because you didn't press 2 to save your re-recorded greeting… Well that's wrong. Your existing greeting gets overridden as soon as you start the recording. Even if you hang up the phone in the middle of your message, it'll still be recorded and played as your greeting. I don't think that this calls for a workaround or more studying of the app. I am using the default settings on this and think that it can be improved, so that you can review your greeting before committing it with a save action. Being able to listen to your existing greeting before re-recording it wouldn't be a bad add-on either. What is your take on this? Emrah

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is the example flex client can't receive a call?
by siklub 19 Sep '12

19 Sep '12

Hi All, I just try the example flex client. It can make a call to 888(a)conference.freeswitch.org But it can't receive a call. Some error message, maybe: switch_ivr_originate.c:2591 Cannot create outgoing channel of type [error] cause: [USER_NOT_REGISTERED] mod_dptools.c:3027 Originate Failed. Cause: USER_NOT_REGISTERED .... rtmp.c:100 [amfnumber=2] Unhandled control packet (type=0x3) Is the example can receive a call? (it is only example, maybe it is incomplete).

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no audio from portaudio in conference
by Vik Killa 19 Sep '12

19 Sep '12

I have audio playing into the MIC (capture) on my sound card. The device is "/dev/dsp" I know there is audio being played into the sound card because when I run: sox -t ossdsp -w -s /dev/dsp ./tmp.wav and listen to tmp.wav, I hear the audio. When I call a conference using "/dev/dsp" no audio is being played from the capture into the conference. If I use freeswitch-1.06, portaudio seems to work. I'd like to use the latest FS version for many reasons. Any ideas what has changed in mod_portaudio? Maybe something broke? It doesn't seem like many people use this module and may not be maintained or bugs may not be submitted for it. I'll do any debugging necessary and any advice would be appreciated. Thanks.

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freetdm and destination_number
by Tim Bock 18 Sep '12

18 Sep '12

Hello, I'm rather new with FreeSWITCH, migrating over from an asterisk setup. I have a Digium TDM card, with 3 fxo ports and 1 fxs port. I have everything mostly working, but I'm having an issue with incoming fxo calls. I've found examples where people have used destination_number in their condition, but this doesn't seem to work for me. I can successfully use > <condition field="${channel_name}" expression="^(FreeTDM/2:3/)"> but not > <condition field="destination_number" expression="(7832889)$"/> The issue seems to be that destination_number is set to 2 early on, somehow: > 012-08-23 20:40:43.213780 [DEBUG] ftmod_analog.c:646 [s2c3][1:3] > Executing stat > e handler on 2:3 for RING > 2012-08-23 20:40:43.213780 [DEBUG] mod_freetdm.c:1992 got FXO sig 2:3 > [START] > 2012-08-23 20:40:43.213780 [DEBUG] ftdm_io.c:3131 [s2c3][1:3] Enabled > software D > TMF detector > 2012-08-23 20:40:43.213780 [DEBUG] mod_freetdm.c:407 Set codec PCMU 20ms > 2012-08-23 20:40:43.213780 [DEBUG] mod_freetdm.c:1740 Connect inbound > channel Fr > eeTDM/2:3/2 so that later: > Dialplan: FreeTDM/2:3/2 Regex (FAIL) [fxo_incoming] > destination_number(2) =~ /(78328 > 89)$/ break=on-false Any ideas on what I am doing wrong? I can make it work using the channel_name variable, but using destination_number would be much more convenient. Thank you, Tim

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Strip digits from incoming caller id
by georg＠riseup.net 18 Sep '12

18 Sep '12

Hi all, My provider is sending me the phone numbers in the E.164 (I guess) format. I would like to remove the international prefix (on local fixed line and local mobile calls at least), and add a leading zero. I know how to add digits, but I've got problems to remove digits. My dialplan at the moment looks like this: <extension name="Dispo Intern" continue="true"> <condition field="destination_number" expression="^(123456789)$"> <regex field="caller_id_number" expression="^49(\d+)$2"/> <action application="set" data="effective_caller_id_number=0$2"/> <action application="set" data="effective_caller_id_name=_undef_"/> <anti action application="set" data="effective_caller_id_number=${original_caller_id_number}"/> </condition> </extension> What I'm trying to do: - Check if the call is a "local" one - If yes, alter the caller id - I thought, I'm able to save the stripped caller id in $2, and use it again in the next line; this is not true, or at least not in the way I did this - It seems that $2 is "empty", because the altered caller id is just 0 Could someone point me into the right direction? Thanks in advance, Georg

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xml curl
by Miha 18 Sep '12

18 Sep '12

hi, is it possible to use xml_curl just for a one part of my dialplan? Regards, Miha

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t.38 re-invite results in 407 Proxy Authentication Required
by Christian Löschenkohl 18 Sep '12

18 Sep '12

hello fs users anybody have similar problems than me? symptom is that within a t.38 re-invite freeswitch responds with 407 Proxy Authentication Required wasn't the case in e.g. 1.2.0-rc2 now with 1.2.2+git~20120911T185917Z~3fd1a57902 it's like described was this an intended changed. if yes - why? many sip ua devices only respond with an ACK instead of an re-invite with user credentials. br -- Ing. Christian Löschenkohl Senior VoIP Engineer Head of VoIP Engineering Head of R&D T +43 5 77 11 - 1000 F +43 5 77 11 - 1002 E christian.loeschenkohl(a)xpirio.com W www.xpirio.com xpirio Telekommunikation & Service GmbH Gerichtsstand Klagenfurt FN297465z Lakeside B04 | 9020 Klagenfurt | Austria An ESS Group company

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