March 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Re: [Freeswitch-users] Questions on Building an application for FreeSWITCH
by Kristian Kielhofner 09 Jun '12

09 Jun '12

Glad to hear. I still say to build the app and logic completely separately and connect it to FreeSWITCH over the socket using ESL. -- Kristian Kielhofner http://blog.krisk.org ----- Original Message ----- From: freeswitch-users-bounces(a)lists.freeswitch.org <freeswitch-users-bounces(a)lists.freeswitch.org> To: freeswitch-users(a)lists.freeswitch.org <freeswitch-users(a)lists.freeswitch.org> Sent: Wed May 12 18:28:13 2010 Subject: Re: [Freeswitch-users] Questions on Building an application forFreeSWITCHThis is legit. We are not into harassing anyone. There are too many people doing that already. We currently have this working on a Windows/Dialogic platform, but would like to get it over to FreeSWITCH for a number of reasons. It is always nice to avoid spending alot of time developing using a method that you later find has serious issues or limitations and then having to change horses and start over. I just thought I would tap on someone else's wisdom in regards to a good way to talk to FS that works well. I did notice that the wiki talks like LUA is the preferred way to go. Thanks Michael Collins wrote: > > > On Wed, May 12, 2010 at 2:50 PM, Kristian Kielhofner > <kris(a)kriskinc.com <mailto:kris@kriskinc.com>> wrote: > > Are you going to be calling my cell phone to ask if I owe $10,000 > or more to > the IRS? > > Seriously, the world does not need another robodialer to harass > people... > > Agreed, however there are legitimate use cases for this, like a > retailer calling to say an order is in, or reminder calls sent out by > a doctor's office. But yes, if it's dialing for dollars then I'm with > you: Por favor! No mas! > -MC > > > > With that being said, write your app in whatever you want using > ESL and > connect to the socket to originate calls and listen for events. > > > -- > Kristian Kielhofner > http://blog.krisk.org > > ----- Original Message ----- > From: freeswitch-users-bounces(a)lists.freeswitch.org > <mailto:freeswitch-users-bounces@lists.freeswitch.org> > <freeswitch-users-bounces(a)lists.freeswitch.org > <mailto:freeswitch-users-bounces@lists.freeswitch.org>> > To: freeswitch-users(a)lists.freeswitch.org > <mailto:freeswitch-users@lists.freeswitch.org> > <freeswitch-users(a)lists.freeswitch.org > <mailto:freeswitch-users@lists.freeswitch.org>> > Sent: Wed May 12 15:40:58 2010 > Subject: [Freeswitch-users] Questions on Building an application > forFreeSWITCHI am looking for a boot in the right direction on the > following project. > > I am looking to build an "application" with FS where it will > frequently > check a database and then make an outbound call when a record with > certain conditions is found in the DB. It needs to determine if the > call was then answered by a machine/vm, busy, or answered by a person, > play a message and get a response back to be recorded in the db. > It may > find many calls that need to be made at the same time. It will be > running on a PRI. > > I am confused as to what approach to use to accomplish this. > > Lua Script, Perl Script, Javascript, Phython, PHP, Ruby....etc. > Liverpie, ESL....etc. > Or a totally different approach. > > I realize that there is not only one "correct" way to build this, > but I > am looking for advise from those who have done this type of thing > before. I value the opinion of those who have gone down this road and > would be willing to share their thoughts on a recommended path to take > to accomplish this. > > Thanks > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org > <mailto:FreeSWITCH-users@lists.freeswitch.org> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org > <mailto:FreeSWITCH-users@lists.freeswitch.org> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > ------------------------------------------------------------------------ > > _______________________________________________ > 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 > _______________________________________________ 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

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Facebook...
by Klaus Hochlehnert 27 May '12

27 May '12

Hi Guys, just saw that you already have a Facebook entry. Wouldn't it be possible to connect the news/twitter feeds somehow with Facebook? I'd really like to see all news on Facebook... Cheers, Klaus

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Choopy one-way noise
by Christian Benke 11 May '12

11 May '12

Hello! On a box with a Sangoma A500 BRI card i have a serious audio quality problem. The "outside"-leg of a call hears a choppy robotic noise, it's difficult to understand the actual voice. I hope someone recognizes this issue, as i have tried several approaches and so far i neither know what causes the problem nor where it is caused exactly. Here's a sample file(The noise is not very noticeable, it's the oscillating hum you hear in the background, the short voice-sample you hear is mine, but it's inbound so it's not affected): http://poab.org/test1.wav If the call is set on-hold or if there's a playback/moh, the quality for the outside-leg is fine while there's no two-way-audio, so my assumption was that it's some kind of problem with codecs or transcoding. Capturing the RTP-Stream from the phone to the server proved though that the quality is fine before it arrives at FS(There are no audio problems on internal calls as well). Here's what else i've tried without success: - set disable-monotonic-timing to true - started FS with the -rt Flag(As well as -waste and -heavy-timer) - jitter "time_test" 100 100 responds with avg 130(Thats fine right?) - rearranged the codec-settings of the phone(Sangoma SPIP 450) to give A-law the highest priority - high-precision-timer is available in the kernel according to dmesg and /proc/timer_list (kernel is 2.6.32-5-686 on debian 6.0) Sangoma's Support appears to be clueless as well, they didn't have effective suggestions so far(Opened a ticket 2 weeks). Since i don't have this problem with identical hardware without Sangoma-Cards(SIP-Trunk instead), my first assumption was the card. Probably an interrupt issue? Any suggestions would be highly appreciated, i'm a bit at loss now and only know more drastic next steps(Replacing ISDN-Modules or the whole hardware). Best regards, Christian

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RPMs and Yum repo for EL5 and EL6 based system in beta
by Ken Rice 19 Apr '12

19 Apr '12

Hey guys, Check it out... Yum is on the way for new freeswitch goodness. With root perms (that means logged in as root, su or maybe even sudo) rpm -Uvh http://files.freeswitch.org/yum/freeswitch-release-1-0.noarch.rpm This will install the FreeSwitch YUM repo on your RHEL5 or 6 based box (that includes Centos or Scientific Linux 5 and 6 platforms) Then to install FreeSWITCH with the same configs you get from making samples yum install freeswitch-config-vanilla This will install freeswitch to /usr/local/freeswitch as you are use to doing at this time. RPMs are being rolled every couple of days right now, but will soon be rolled nightly... Find me on #freeswitch on irc.freenode.net as SwK if you have any questions or issues, if you find any bugs please open a Jira and assign it to me. These RPMs are Alpha/Beta quality please test them before using them in a production environment... Additional Packages will be available shortly. Ken aka SwK

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SIP trace specific call or uuid
by Anton VG 18 Apr '12

18 Apr '12

HI! Is there a way to enable sip-trace on the specific call or uuid? I've been only able to find how to enable it per profile, but on a busy situation, the information amount is too high to be useful, so it's often necessary to trace a specific call or call to specific destination.

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Open Bugs on Jira and Call for help
by Ken Rice 11 Apr '12

11 Apr '12

If you have open bugs on Jira, please make sure they are up to date on any all information.... If you wish to address a specific bug, please visit us on the weekly conference call so we can address them... We¹re getting close to time for rolling Release Candidate for 1.2 and we want to address as many of these as possible... Also, I¹m looking for some configuration documentation help. If you¹ve pulled git in the past few days you¹ll notice that in the module src directories there is now a conf directory, we need some help with making sure all the configuration options are documented and what their default settings are. The end goal is to get all the options documented in their default states. What skills do you need? The ability to read a little C, and the ability to use a text editor to edit/create configs... K

2 6

Strange DTMF Tones On Inbound Calls
by Daniel Knaggs 10 Apr '12

10 Apr '12

Hello all, Got a bit of a strange one, we appear to be getting DTMF tones on incoming calls when the caller hasn’t even pressed any keys. It normally happens with 10 seconds or so after the call has been answered. Here is the log of it happening earlier: - 2012-03-23 10:38:16.852654 [DEBUG] ftdm_io.c:3530 [s1c2][1:2] Queuing DTMF A (debug = 0) 2012-03-23 10:38:16.852654 [DEBUG] mod_freetdm.c:799 Queuing DTMF [A] in channel FreeTDM/1:2/000 device 1:2 2012-03-23 10:38:16.915653 [DEBUG] switch_rtp.c:2420 Send start packet for [A] ts=49440 dur=160/160/2000 seq=46803 lw=49440 2012-03-23 10:38:16.936653 [DEBUG] switch_rtp.c:2323 Send middle packet for [A] ts=49440 dur=320/320/2000 seq=46804 lw=49600 2012-03-23 10:38:16.957652 [DEBUG] switch_rtp.c:2323 Send middle packet for [A] ts=49440 dur=480/480/2000 seq=46805 lw=49760 2012-03-23 10:38:16.978652 [DEBUG] switch_rtp.c:2323 Send middle packet for [A] ts=49440 dur=640/640/2000 seq=46806 lw=49920 2012-03-23 10:38:16.999652 [DEBUG] switch_rtp.c:2323 Send middle packet for [A] ts=49440 dur=800/800/2000 seq=46807 lw=50080 2012-03-23 10:38:17.020651 [DEBUG] switch_rtp.c:2323 Send middle packet for [A] ts=49440 dur=960/960/2000 seq=46808 lw=50240 2012-03-23 10:38:17.041651 [DEBUG] switch_rtp.c:2323 Send middle packet for [A] ts=49440 dur=1120/1120/2000 seq=46809 lw=50400 2012-03-23 10:38:17.062651 [DEBUG] switch_rtp.c:2323 Send middle packet for [A] ts=49440 dur=1280/1280/2000 seq=46810 lw=50560 2012-03-23 10:38:17.083650 [DEBUG] switch_rtp.c:2323 Send middle packet for [A] ts=49440 dur=1440/1440/2000 seq=46811 lw=50720 2012-03-23 10:38:17.104650 [DEBUG] switch_rtp.c:2323 Send middle packet for [A] ts=49440 dur=1600/1600/2000 seq=46812 lw=50880 2012-03-23 10:38:17.125650 [DEBUG] switch_rtp.c:2323 Send middle packet for [A] ts=49440 dur=1760/1760/2000 seq=46813 lw=51040 2012-03-23 10:38:17.146650 [DEBUG] switch_rtp.c:2323 Send middle packet for [A] ts=49440 dur=1920/1920/2000 seq=46814 lw=51200 2012-03-23 10:38:17.167661 [DEBUG] switch_rtp.c:2323 Send end packet for [A] ts=49440 dur=2080/2080/2000 seq=46815 lw=51200 2012-03-23 10:38:17.167661 [DEBUG] switch_rtp.c:2323 Send end packet for [A] ts=49440 dur=2080/2080/2000 seq=46816 lw=51200 2012-03-23 10:38:17.167661 [DEBUG] switch_rtp.c:2323 Send end packet for [A] ts=49440 dur=2080/2080/2000 seq=46817 lw=51200 2012-03-23 10:38:17.167661 [DEBUG] switch_rtp.c:2271 Queue digit delay of 40ms 2012-03-23 10:38:18.070638 [DEBUG] ftdm_io.c:3530 [s1c2][1:2] Queuing DTMF B (debug = 0) 2012-03-23 10:38:18.070638 [DEBUG] mod_freetdm.c:799 Queuing DTMF [B] in channel FreeTDM/1:2/000 device 1:2 2012-03-23 10:38:18.133637 [DEBUG] switch_rtp.c:2420 Send start packet for [B] ts=59040 dur=160/160/2000 seq=46864 lw=59040 2012-03-23 10:38:18.154637 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=59040 dur=320/320/2000 seq=46865 lw=59200 2012-03-23 10:38:18.175637 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=59040 dur=480/480/2000 seq=46866 lw=59360 2012-03-23 10:38:18.196636 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=59040 dur=640/640/2000 seq=46867 lw=59520 2012-03-23 10:38:18.217636 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=59040 dur=800/800/2000 seq=46868 lw=59680 2012-03-23 10:38:18.238636 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=59040 dur=960/960/2000 seq=46869 lw=59840 2012-03-23 10:38:18.259636 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=59040 dur=1120/1120/2000 seq=46870 lw=60000 2012-03-23 10:38:18.280636 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=59040 dur=1280/1280/2000 seq=46871 lw=60160 2012-03-23 10:38:18.301635 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=59040 dur=1440/1440/2000 seq=46872 lw=60320 2012-03-23 10:38:18.322635 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=59040 dur=1600/1600/2000 seq=46873 lw=60480 2012-03-23 10:38:18.343634 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=59040 dur=1760/1760/2000 seq=46874 lw=60640 2012-03-23 10:38:18.364634 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=59040 dur=1920/1920/2000 seq=46875 lw=60800 2012-03-23 10:38:18.385634 [DEBUG] switch_rtp.c:2323 Send end packet for [B] ts=59040 dur=2080/2080/2000 seq=46876 lw=60800 2012-03-23 10:38:18.385634 [DEBUG] switch_rtp.c:2323 Send end packet for [B] ts=59040 dur=2080/2080/2000 seq=46877 lw=60800 2012-03-23 10:38:18.385634 [DEBUG] switch_rtp.c:2323 Send end packet for [B] ts=59040 dur=2080/2080/2000 seq=46878 lw=60800 I’m very sure the caller isn’t dialling “A” or “B”! Running on ISDN via a TE121 card using E1 (euroisdn). Freeswitch version is “FreeSWITCH Version 1.0.head (git-b9b7266 2012-02-10 12-23-58 -0600)”. Currently there is a “bind_meta_app” in the config which binds a script on the B leg of the call which parks the call – I haven’t tried turning this off yet to see if it’s this. Wondering if anyone has any ideas or has come across this before? Thanks in advance. [cid:imageacd695.PNG@9e92e461.40aa4b96] Daniel Knaggs Software Developer Reality Solutions Ltd, 1 Global Business Park, Hamburg Road, Kingston upon Hull, East Yorkshire, HU7 0AE Tel: 01482 828000 / Fax: 01482 373100 Daniel.Knaggs(a)realitysolutions.co.uk<mailto:Daniel.Knaggs@realitysolutions.co.uk> www.realitysolutions.co.uk<http://www.realitysolutions.co.uk/> ________________________________ Sage Accredited Business Partner serving businesses in Yorkshire & Lincolnshire [cid:image27a71e.PNG@c2da8488.4683bff1]

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Acoustic Echo Cancellation on Freeswitch side
by Imran Moinuddin 10 Apr '12

10 Apr '12

Hi there, I was wondering whether there were any echo cancellation alternatives available that could potentially work on the Freeswitch server side. This is mainly to cater for people that dial into my service using just their laptop microphone + speakers and their client app doesn't support echo cancellation. I realize that this could be an expensive proposition (from a computational perspective) but curious if the option (commercial or otherwise) exists. Thanks! Imran

7 6

Dialplan/PHP esl variable problem.
by Alex Crow 09 Apr '12

09 Apr '12

Hi all, I have a dialplan that calls a PHP script through CURL: <extension name="test" continue="false"> <condition field="destination_number" expression="^669$"> <action application="answer"/> <action application="sleep" data="1000"/>  <action application="curl" data="http://localhost/provision/index.php?action=test&uuid=${uuid}&exten=${calle…" /> <action application="sleep" data="1000"/> <action application="log" data="INFO ${curl_response_data}"/> <action application="hangup" data="NORMAL_CLEARING"/> </condition> </extension> (notice the play_and_get_digits is commented out) and in PHP I connect via ESL via this function and code: function saySomething($uuid) { $sock = new ESLconnection('localhost','8021','ClueCon'); sleep (2); $res = $sock->execute("play_and_get_digits","4 11 1 5000 # ivr/ivr-please_enter_the_phone_number.wav ivr/ivr-that_was_an_invalid_entry.wav target_num \d+", "$uuid"); //$foo = $uuid . ' target_num'; sleep (2); $res=$sock->sendRecv("api uuid_getvar $uuid target_num"); $ret = $res->getBody(); $sock->disconnect(); //echo $ret; return ($ret); } if ($action == 'test') { $res = saySomething($_REQUEST['uuid']); echo $res; } However if I run the play_and_get_digits from the PHP the subsequent uuid_getvar returns _undef_. If I comment out the play_and_get_digits in PHP and uncomment it in the dialplan, the uuid_getvar *does* return the digits pressed. Is there something I'm doing wrong here? Thanks Alex -- This message is intended only for the addressee and may contain confidential information. Unless you are that person, you may not disclose its contents or use it in any way and are requested to delete the message along with any attachments and notify us immediately. "Transact" is operated by Integrated Financial Arrangements plc Domain House, 5-7 Singer Street, London EC2A 4BQ Tel: (020) 7608 4900 Fax: (020) 7608 5300 (Registered office: as above; Registered in England and Wales under number: 3727592) Authorised and regulated by the Financial Services Authority (entered on the FSA Register; number: 190856)

5 21

X-Lite and Video
by peter＠uringme.com 04 Apr '12

04 Apr '12

(Apologies to the list owner for originally sending this to the wrong address) I have two laptops running X-Lite 4. I have them registered to a FreeSwitch server (latest git) as extensions 7777 and 7778. I have a dialplan for each (quick and dirty) that just bridges them when one is dialed from the other: <extension name="7777"> <condition field="destination_number" expression="^7777$"> <action application="bridge" data="sofia/external/7777%test"/> </condition> </extension> (and vice-versa for 7778). I can dial between them just fine for audio calls -- bidirectional audio, etc, no problem. I'm trying to get video going. Both X-Lites have H.263 and H.263-1998 enabled in their settings. Freeswitch has the following in vars.xml: <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G7221@32000h,G7221@16000h,G722,PCMU,PCMA,GSM,H263,H264,H263-1998"/> When I try to make a video call from one extension to the other, the calling extension seems to think it's in video, but the called extension doesn't. INVITE from freeswitch console: ------------------------------------------------------------------------ INVITE sip:7778@test SIP/2.0 Via: SIP/2.0/UDP 192.168.1.7:50350;branch=z9hG4bK-d8754z-9ea3618224c5cf23-1---d8754z-;rport Max-Forwards: 70 Contact: <sip:7777@68.202.69.172:50350> To: <sip:7778@test> From: "Peter Test"<sip:7777@test>;tag=a7031d83 Call-ID: ZDUzZGE1YjUyOTQ2ZGNmZTY0Yjc5ODA5NTE4NDAzMGQ. CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp Supported: replaces User-Agent: X-Lite 4 release 4.1 stamp 63214 Content-Length: 681 v=0 o=- 12970176286658638 1 IN IP4 192.168.1.7 s=CounterPath X-Lite 4.1 c=IN IP4 192.168.1.7 t=0 0 a=ice-ufrag:20fef6 a=ice-pwd:0a03863684bc5f16a9c862dcdccdd8eb m=audio 58632 RTP/AVP 107 0 8 101 a=rtpmap:107 BV32/16000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=candidate:1 1 UDP 659136 192.168.1.7 58632 typ host a=candidate:1 2 UDP 659134 192.168.1.7 58633 typ host m=video 50994 RTP/AVP 34 115 a=rtpmap:34 H263/90000 a=fmtp:34 QCIF=2;CIF=2;VGA=2 a=rtpmap:115 H263-1998/90000 a=fmtp:115 QCIF=2;CIF=2;VGA=2;I=1;J=1;T=1 a=sendrecv a=candidate:1 1 UDP 659136 192.168.1.7 50994 typ host a=candidate:1 2 UDP 659134 192.168.1.7 50995 typ host I do see freeswitch seeing the audio and video codecs: 2012-01-04 13:44:37.188657 [DEBUG] sofia_glue.c:4683 Audio Codec Compare [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000] 2012-01-04 13:44:37.188657 [DEBUG] sofia_glue.c:2800 Set Codec sofia/external/7777@test PCMU/8000 20 ms 160 samples 64000 bits 2012-01-04 13:44:37.188657 [DEBUG] switch_core_state_machine.c:343 (sofia/external/7777@test) State NEW 2012-01-04 13:44:37.188657 [DEBUG] sofia_glue.c:4797 Set 2833 dtmf send/recv payload to 101 2012-01-04 13:44:37.188657 [DEBUG] sofia_glue.c:4856 Video Codec Compare [H263:34]/[H263:34] However, when freeswitch starts the bridge and calls the far-end party, it doesn't send along video information in the INVITE: INVITE sip:7778@68.202.69.172:32834;transport=udp;rinstance=677b87c43ee7970a SIP/2.0 Via: SIP/2.0/UDP 204.13.175.89:5080;rport;branch=z9hG4bK8jm56tcmZ6p6j2012-01-04 Max-Forwards: 69 From: "Peter Test" <sip:7777@204.13.175.89>;tag=ZeZUav5XXat5e To: <sip:7778@68.202.69.172:32834;transport=udp;rinstance=677b87c43ee7970a> Call-ID: fe64b2f5-b1a6-122f-a187-00144f49eecc CSeq: 22515274 INVITE Contact: <sip:mod_sofia@204.13.175.89:5080> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-1086cba 2011-05-23 22-51-43 -0500 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 207 X-FS-Support: update_display Remote-Party-ID: "Peter Test" <sip:7777@204.13.175.89>;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1325690959 1325690960 IN IP4 204.13.175.89 s=FreeSWITCH c=IN IP4 204.13.175.89 t=0 0 m=audio 11718 RTP/AVP 0 8 3 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 So, am I missing something? Do I need to use something other than "bridge" in the dialplan, or do I need to add some variables to be able to pass on the video? All I'm trying to do is make a video call between two X-Lites that are locally SIP registered to freeswitch. Because I want to record the video at some point in the future, I don't want to divert the media -- I want it streaming/passing through freeswitch. When the call is connected, the caller shows a "Waiting for video", but the called doesn't show this. When I try to start the video, it says "Failed to Start Video".

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