February 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

EVENT_TRAP: IP change detected -> no Write unlock -> Error Creating SIP UA for profile: internal
by Nestor A Diaz 31 Jul '12

31 Jul '12

Hi Guys. Today my standard sip profile decide not to work anymore, in fact it was related with the fact i had put some ip alias on the same internal network interface a few days ago, by searching the logs, this is what i got: 2012-02-28 14:50:56.126576 [INFO] mod_sofia.c:5155 EVENT_TRAP: IP change detected 2012-02-28 14:50:56.126576 [INFO] mod_sofia.c:5156 IP change detected []->[] [::1]->[fe80::225:90ff:fe3a:2e1a] 2012-02-28 14:50:56.306534 [NOTICE] sofia_glue.c:5295 Reload XML [Success] [...] 2012-02-28 14:50:57.926576 [DEBUG] sofia.c:2121 Write lock doublenat 2012-02-28 14:50:57.926576 [DEBUG] sofia.c:2132 Write unlock doublenat [...] 2012-02-28 14:50:58.026533 [DEBUG] sofia.c:2121 Write lock internal-ipv6 2012-02-28 14:50:58.026533 [DEBUG] sofia.c:2132 Write unlock internal-ipv6 [...] 2012-02-28 14:50:58.026533 [ERR] sofia.c:1916 Error Creating SIP UA for profile: internal-ipv6 [...] 2012-02-28 14:50:58.026533 [DEBUG] sofia.c:2121 Write lock internal but no unlock internal profile and no creating sip ua for profile: internal. Later final users reported the problem and i tried to start the profile: $ sofia profile internal start and i got: 2012-02-28 17:08:30.586507 [ERR] sofia.c:1916 Error Creating SIP UA for profile: internal So i deactivated the ip aliases and restart freeswitch and everything comes to live. But I suspect a problem with sofia because it wasn't able to unlock the internal profile, i checked the status of sofia via command line when this happens and there was no sofia internal profile listed. I don't know how the ip change event was triggered since i added the ip alias some days ago, not today when the problem was triggered. Maybe there will be some bug there. Slds. -- Nestor A. Diaz Ingeniero de Sistemas Tel. +57 1-485-3020 x 211 Cel. +57 316-227-3593 Tel. SIP: sip:211@tiendalinux.com Email/MSN: nestor(a)tiendalinux.com http://www.tiendalinux.com/ Bogota, Colombia

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How to break a playback with loops
by Daniel Liang 24 Jun '12

24 Jun '12

Hi, I am trying to playing back a wav file a few times, and on receiving a dtmf event, it stops playing the file. The command I sent is: sendmsg call-command: execute execute-app-name: playback execute-app-arg: <the wav file> loops: 4 On receiving a dtmf event, I send a break command: sendmsg call-command: execute execute-app-name: break However, that only breaks one loop of the playback. And I have to press a dtmf for 4 times to stop the file. I tried sending 4 break commands all at once. But it didn't work. And even worse, my program stops receiving dtmfs anymore. My guess is that the first break stops the first loop, but before the second loop starts, the following 3 breaks stop something else that I don't know what they are. Any help is appreciated. Thanks. Daniel Liang

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Event Socket
by Roger Castaldo 11 Jun '12

11 Jun '12

Hi everyone i have a simple question, is there an api command that will list all the available events that can be listened on by the event socket? i realize there is a wiki page containing a list but for my purposes an api command to list the available event names would be much better.

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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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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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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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Compiling Latest GIT
by Peter Fraser 28 Mar '12

28 Mar '12

Hi, I am having all sorts of trouble with compiling freeswitch on Ubuntu 11.10. I have downloaded the source from the latest GIT and and followed the wiki. I have used freeswitch many times in the past however I am not the best on compiling with make etc. I am getting an error after i run the initial make. ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:1890:56: error: expected ')' before 'const' ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:1893:57: error: expected ')' before 'const' ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:2945:2: error: #endif without #if nua_subnotref.c: In function 'nua_refer_client_response': nua_subnotref.c:915:7: error: 'sip_refer_sub_t' undeclared (first use in this function) nua_subnotref.c:915:23: error: expected ';' before 'const' nua_subnotref.c:917:11: error: 'rs' undeclared (first use in this function) cc1: all warnings being treated as errors make[9]: *** [nua_subnotref.lo] Error 1 make[8]: *** [all] Error 2 make[8]: *** No rule to make target `nua/libnua.la', needed by ` libsofia-sip-ua.la'. Stop. make[7]: *** [all-recursive] Error 1 Making all in packages make[6]: *** [all-recursive] Error 1 make[5]: *** [all] Error 2 make[4]: *** [/usr/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/ libsofia-sip-ua.la] Error 2 make[3]: *** [mod_sofia-all] Error 1 make[2]: *** [all-recursive] Error 1 make[1]: *** [all-recursive] Error 1 make: *** [all] Error 2 schepisi@gfi-fraser:/usr/src/freeswitch$ There are more errors but this is where it bombs out. Is anyone able to provide any pointers as to what I might be doing wrong. Thanks in advanced. Peter Fraser

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skypopen scalability tests
by Giovanni Maruzzelli 24 Mar '12

24 Mar '12

On Sat, Apr 30, 2011 at 6:52 PM, Anton VG <anton.vazir(a)gmail.com> wrote: >> Any other ideas on how to proceed for scalability tests? >> > > Hm, since 2.0.0.72 still allows simultaneous calls, you do not have to > register lots of usernames, just 2 of them for 2 PC's > than make 100 calls from 1 machine to another I was going that way, couple years ago, but the Skype client was scaling kind of badly for me. No more than a bunch of simultaneus calls. But at that time I was testing using a standard ALSA sound driver, and Skype client for ALSA. Maybe using Skype for OSS and skypopen.ko it will scale better. Good hint, at least I will check into it in the future. Other ideas? -giovanni -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618

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Direct Call Pickup - Issue
by Lloyd Aloysius 21 Mar '12

21 Mar '12

Hi , FreeSWITCH Version 1.0.head (git-0675b59 2011-06-06 21-28-14 -0500) I have a direct call pickup issue / Bug? 1. Extension 201 On the phone 2. Second call Comes to Extension 201 3. Extension 202 Dial **201 4. Extension 201 Lost the Active call and Extension 202 Pickup the Extension 201 Active Call. But Extension 202 should pickup the second call not active call is this a bug? I am using Default Dial plan application  <extension name="intercept-ext"> <condition field="destination_number" expression="^\*\*(\d+)$"> <action application="answer"/> <action application="intercept" data="${hash(select/${domain_name}-last_dial_ext/$1)}"/> <action application="sleep" data="2000"/> </condition> </extension> Thanks in advanced Lloyd

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