January 2009 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Starting FS on OSX
by Martin Joseph 18 Jan '09

18 Jan '09

Hello again FreeSwitchers, I have built the 1.02 on 10.4.11(OSX) and had no problems with that. I have never been able to build from the SVN, but that is another story. Now that I have migrated to 1.02 I was wondering if I can get some help on a long standing issue I have with starting FS at boot. I am hoping to use Launchd which is the standard on OSX 10.4 and I attempted to cobble together a script, but haven't had great results. I did search for wiki entries on this, but haven't found any help with it. Ideas? Thanks, Marty

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Changing internal profile
by kriko 18 Jan '09

18 Jan '09

I'm trying to change internal address to something else than default external, so I would be able to make calls from phone on wireless to e.g. gtalk users. I already tried this scenario at different location and it work, but everything was on same lan, now I have eth0 interface with external ip e.g. 212.235.180.41 eth1 interface with internal ip 192.168.0.1 on eth1 is wireless router attached with wireless clients (my phone is 192.168.0.102), messages are going trough, no problem with that. If I got it right I have to change: <param name="rtp-ip" value="$${local_ip_v4}"/> <param name="sip-ip" value="$${local_ip_v4}"/> to 192.168.0.1, after that I modified acl.conf.xml domains section to: <list name="domains" default="deny"> <node type="allow" domain="$${domain}"/> <node type="allow" domain="192.168.0.1"/> </list> but I'm getting forbidden: 2009-01-03 09:39:42 [WARNING] sofia_reg.c:1533 sofia_reg_parse_auth() Can't find user [1000(a)192.168.0.1] You must define a domain called '192.168.0.1' in your directory and add a user with the id="1000" attribute and you must configure your device to use the proper domain in it's authentication credentials. What directory? Should I delete added node from acl.conf.xml? -- kriko

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[ANN] Spice SoftPhone, a softphone GUI for FreeSWITCH
by Andrew Thompson 18 Jan '09

18 Jan '09

I'd like to announce the first beta release of a cross-platform ruby/tk GUI for using FreeSWITCH like a soft-phone (using mod_portaudio). It's not particularly fancy, but I needed a cross platform softphone with good voice quality that was debuggable and didn't have a ton of features to confuse the users. I couldn't find one so we built one. I've got some sparse documentation up at: http://opencsm.org/wiki/index.php/Spice_SoftPhone And you can download it from http://opencsm.org/download . It's under the MPL and I've been cleared to re-licence my other FreeSWITCH related projects under the MPL too. I've tested it on Windows, FreeBSD, Solaris and OSX (it used to work on linux, I assume it still does). Comments/complaints/bugreports welcome. It's definitely still got some rough spots (I don't think it'll run without a controlling terminal, for example), but we're going to be polishing it up and hopefully putting it in production here in the next few weeks to replace a very buggy closed-source phone we've had to endure far too long. Please download it if you're interested, the download count helps us continue working on this kind of stuff :) Andrew - opencsm.org

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Re: [Freeswitch-users] Dialing Out Problem via Gateway
by Will Smith 16 Jan '09

16 Jan '09

Thank you Brian, The problem is very simple, I or the other party cannot hear each other when I first dial and the other party picks up the phone. We hear the phone ring, the other end picks up the phone says something, but I cannot hear - nothing, even static. Same thing happen on my end, I say something, and the other end do not hear a thing. When I put the call on hold, the other end can hear music on hold. When I take the call back, now we can talk. Something does not go through when the other end picks up the call. This is the extension in the dialplan/default.xml if (typeof YAHOO == "undefined") { var YAHOO = {}; } YAHOO.Shortcuts = YAHOO.Shortcuts || {}; YAHOO.Shortcuts.hasSensitiveText = false; YAHOO.Shortcuts.sensitivityType = []; YAHOO.Shortcuts.doUlt = false; YAHOO.Shortcuts.location = "us"; YAHOO.Shortcuts.document_id = 0; YAHOO.Shortcuts.document_type = ""; YAHOO.Shortcuts.document_title = "t"; YAHOO.Shortcuts.document_publish_date = ""; YAHOO.Shortcuts.document_author = "willbelair(a)yahoo.com"; YAHOO.Shortcuts.document_url = ""; YAHOO.Shortcuts.document_tags = ""; YAHOO.Shortcuts.document_language = "english"; YAHOO.Shortcuts.annotationSet = { "lw_1232141334_0": { "text": "9054516117", "extended": 0, "startchar": 391, "endchar": 400, "start": 391, "end": 400, "extendedFrom": "", "predictedCategory": "", "predictionProbability": "0", "weight": 1, "relScore": 0, "type": ["shortcuts:/us/instance/identifier/fedex_tracking"], "category": ["IDENTIFIER"], "wikiId": "", "relatedWikiIds": [], "relatedEntities": [], "showOnClick": [], "context": "", "metaData": { "verified": "false", "visible": "true" } } }; YAHOO.Shortcuts.headerID = "a9059b1f35336b4363b0c75035b61d07"; <extension name="mygateway"> <condition field="destination_number" expression="^(1{0,1}\d{10})$"> <action application="set" data="effective_caller_id_number=12223334444"/>   <action application="bridge" data="sofia/gateway/mygw/$1"/> </condition> </extension> --- On Fri, 1/16/09, Brian West <brian(a)freeswitch.org> wrote: From: Brian West <brian(a)freeswitch.org> Subject: Re: [Freeswitch-users] Dialing Out Problem via Gateway To: freeswitch-users(a)lists.freeswitch.org Date: Friday, January 16, 2009, 1:13 PM Can you detail your problem a bit more? /b On Jan 16, 2009, at 3:09 PM, Will Smith wrote: Hi, I got a strange problem that I don't really understand, and I hope that you could give me some hint how to fix that: When I dial out through a gateway that is defined in the sip_profiles/external , (The xml file is simple as below. ) I cannot talk or hear from the other end. But when I put the line on hold, two ends can hear music, and when open the line again, this time 2 ends can hear and talk. Is there any where that I can fix this problem? Thank you <include> <gateway name="mygw"> <param name="username" value="myusername"/> <param name="realm" value="my_sip_server.com"/> <param name="password" value="mypassword"/> <param name="proxy" value="my_sip_server.com"/> <param name="expire-seconds" value="600"/> <param name="register" value="true"/> </gateway> </include> _______________________________________________ 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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Vmware voice quality
by Remko Kloosterman 16 Jan '09

16 Jan '09

Hello Ken, hello all, I just read about the FreeSWITCH VMware applicance. I'm curious about your experiences with the audio quality on VMWare, so here's a new thread. I've installed freeswitch on VMware Server for Windows. The IVR audio always plays choppy, while the server itself has no performance issues. The same poor voice quality also goes for Asterisk or Yate, even on a very fast VMware ESX system. Did you experience the same and/or do you have pointers on how to troubleshoot and fix this? Thanks, Remko

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Dialing Out Problem via Gateway
by Will Smith 16 Jan '09

16 Jan '09

Hi, I got a strange problem that I don't really understand, and I hope that you could give me some hint how to fix that: When I dial out through a gateway that is defined in the sip_profiles/external , (The xml file is simple as below. ) I cannot talk or hear from the other end. But when I put the line on hold, two ends can hear music, and when open the line again, this time 2 ends can hear and talk. Is there any where that I can fix this problem? Thank you <include> <gateway name="mygw"> <param name="username" value="myusername"/> <param name="realm" value="my_sip_server.com"/> <param name="password" value="mypassword"/> <param name="proxy" value="my_sip_server.com"/> <param name="expire-seconds" value="600"/> <param name="register" value="true"/> </gateway> </include>

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Re: [Freeswitch-users] Sending SMS to SIPtoGSM gateway
by Imthiyaz Ahmed 16 Jan '09

16 Jan '09

Hi This is what I found in the manual SMS and MMS The GSM and CDMA modules can send and receive SMS text messages and MMS multimedia messages to or from a mobile phone. These are transmitted over the Ethernet port as SIP MESSAGE messages to ensure compatibility with a variety of different SIP PBXs. You determine for each module the destination of received SMS or MMS messages and you can program that SMS and MMS messages are always sent to a person who sent an SMS or MMS to that phone number within a specified time. The GS8 modular gateway can send and receive text messages in 7 bit mode (160 characters per message) or in Unicode (70 characters per message). It can send and receive multimedia messages comprising images and rich text that are limited to 30 KB. The sending or receiving client is responsible for formatting or interpreting the message to OMA specifications. Thanks

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Channel variable 'call_timeout'.
by henkoegema 16 Jan '09

16 Jan '09

I use following diaplan for extension 2005: --------------------------------------------------------- <extension name="2005"> <condition field="destination_number" expression="^2005$"> <action application="set" data=”call_timeout=10"/> <action application="set" data="continue_on_fail=USER_BUSY,NO_ANSWER"/> <action application="bridge" data="sofia/internal/2005%$${domain}"/> <action application="voicemail" data="default $${domain} 2005"/> </condition> </extension> What I can't get to work is the "call_timeout" variable. It is set to value 10 at the moment. So I was thinking, that if exten. 2005 is busy or doesn't answer, the call would be transferred to the voicemail after 10 seconds. It is actually transferred after 66 (?) seconds. Changing the value of call_timeout doesn't seem to have any effect. :confused: -- View this message in context: http://www.nabble.com/Channel-variable-%27call_timeout%27.-tp20677406p20677… Sent from the Freeswitch-users mailing list archive at Nabble.com.

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FS doesn't maitain PRI- D-channel state right
by Helmut Kuper 16 Jan '09

16 Jan '09

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, I found, that FS doesn't maintain D-Channel's state correctly. I have a PRI with disabled layer 2 and 3 on TDM side. When FS starts up I get this on console: 2009-01-16 08:16:10 [DEBUG] ozmod_isdn.c:1441 zap_isdn_run() ISDN thread starting. 2009-01-16 08:16:10 [DEBUG] Span:0 Q.921() Starting trunk 0xae411008 (sapi: 0, tei: 0, mode: PTP TE) 2009-01-16 08:16:10 [DEBUG] Span:0 Q.921() Sending SABME 2009-01-16 08:16:10 [DEBUG] Span:0 Q.921() Sending frame - ----------------- Q.921 Packet [Outgoing] --------------- SAPI: 0, TEI: 0, C/R: Command (0) Type: U Frame (SABME) P/F: 1 Q.921 state: "TEI Assigned" (4) [flags: ----] - ---------------------------------------------- 2009-01-16 08:16:10 [CONSOLE] switch_loadable_module.c:857 switch_loadable_module_load_file() Successfully Loaded [mod_openzap] 2009-01-16 08:16:10 [NOTICE] switch_loadable_module.c:141 switch_loadable_module_process() Adding Endpoint 'openzap' 2009-01-16 08:16:10 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'disable_ec' 2009-01-16 08:16:10 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'oz' 2009-01-16 08:16:10 [DEBUG] Span:0 Q.921() New packet received (3 bytes) - ----------------- Q.921 Packet [Incoming] --------------- SAPI: 0, TEI: 0, C/R: Response (0) Type: U Frame (DM (Disconnected Mode)) P/F: 1 Q.921 state: "TEI Assigned" (4) [flags: ----] - ---------------------------------------------- So TDM side tells FS that the link is down: but when I check d-channel's state FS tells me that it is UP: freeswitch@ippbx-prod-node0> oz dump 1 16 API CALL [oz(dump 1 16)] output: span_id: 1 chan_id: 16 physical_span_id: 1 physical_chan_id: 16 type: DQ921 state: UP last_state: DOWN cid_date: cid_name: cid_num: ani: aniII: dnis: rdnis: cause: NONE When I do an outbound call FS throws no error or warning that the link isn't up and gives the call a try. regards helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.9 (MingW32) iEYEARECAAYFAklwNnsACgkQ4tZeNddg3dzjKQCbBDU/SSOyKbD2JGcJFOJZDyBQ nI0An3CFp9HIuTB0cQWT0iJ1Rlx1+yGk =ycwj -----END PGP SIGNATURE-----

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Re: [Freeswitch-users] mod_opal first unsuccessful test
by Apostolos Pantsiopoulos 16 Jan '09

16 Jan '09

Any news regarding this issue? Apostolos Pantsiopoulos wrote: > I am attaching the wireshark capture. Openphone is on xxx.xxx.xxx.202 and > FS is on xxx.xxx.xxx.212 > > Robert Jongbloed wrote: >> >> Can you send me a WireShark capture? >> >> >> >> Robert Jongbloed >> >> OPAL/OpenH323/PTLib Architect and Co-founder. >> >> >> >> *From:* Apostolos Pantsiopoulos [mailto:regs@kinetix.gr] >> *Sent:* Tuesday, 13 January 2009 11:05 PM >> *To:* freeswitch-users(a)lists.freeswitch.org >> *Cc:* Robert Jongbloed >> *Subject:* Re: [Freeswitch-users] mod_opal first unsuccessful test >> >> >> >> I also tried using Ekiga - which is OPAL based - and got the same >> behavior. No audio - although I can see RTP packets. >> >> Apostolos Pantsiopoulos wrote: >> >> Hi, >> >> Yes, openPhone is working with my soundcard. I am using it >> every day for testing purposes. I use the 1.8.1 version. Is there a newer >> version that uses OPAL? I didn't know that. Where can I get it from? >> >> Robert Jongbloed wrote: >> >> Hi guys, >> >> >> >> I was using the OpenPhone that you build with OPAL for my testing. So >> that is identical (I think) to you. >> >> >> >> I have not (yet) do any third party client testing. >> >> >> >> Two ALERTING messages are fine, perfectly legal and OPAL can handle it. >> >> >> >> You say you can see the RTP packets flowing so that implies that the >> mod_opal is actually working, so let's look somewhere else. Have you >> confirmed that OpenPhone is using the sound card correctly? Made a >> call between two machines JUST using OpenPhone for example? >> >> >> >> >> >> Robert Jongbloed >> >> OPAL/OpenH323/PTLib Architect and Co-founder. >> >> >> >> *From:* Anthony Minessale [mailto:anthony.minessale@gmail.com] >> *Sent:* Tuesday, 13 January 2009 6:20 AM >> *To:* Robert Jongbloed >> *Subject:* Fwd: [Freeswitch-users] mod_opal first unsuccessful test >> >> >> >> Heh, >> what client are you using in your tests that are working? >> >> >> ---------- Forwarded message ---------- >> From: *Apostolos Pantsiopoulos* <regs(a)kinetix.gr >> <mailto:regs@kinetix.gr>> >> Date: Mon, Jan 12, 2009 at 9:51 AM >> Subject: [Freeswitch-users] mod_opal first unsuccessful test >> To: freeswitch-users(a)lists.freeswitch.org >> <mailto:freeswitch-users@lists.freeswitch.org> >> >> >> Hi, >> >> I successfully compiled mod_opal using the latest svn for both opal >> and ptlib as Brian suggested. >> >> When I try to establish a call using h323 from my openphone client >> I get no audio although I can see RTP packets in both directions when >> I am >> doing a capture. Some details : >> >> I am using the 11094 revision of the FS trunk. >> I am using the PCMU codec. >> I tried dialing the default IVR (5000) and other testing extensions >> (freeswitch conference, echo test etc.) >> I tried using fast start on and off , h245 tunneling on and off, h245 in >> SETUP on and off. >> >> In my captures I have also noticed a strange behavior : FS sends to >> my client 2 "alerting" packets >> for no apparent reason. Could this be a cause of the problem? >> >> Had anyone any success with mod_opal lately? If yes, could you >> please reply quoting your config >> options (both on FS and on your client)? >> >> >> -- >> ------------------------------------------- >> Apostolos Pantsiopoulos >> Kinetix Tele.com R & D >> email: regs(a)kinetix.gr <mailto:regs@kinetix.gr> >> ------------------------------------------- >> >> >> _______________________________________________ >> 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 >> >> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> >> AIM: anthm >> MSN:anthony_minessale@hotmail.com >> <mailto:MSN%3Aanthony_minessale@hotmail.com> >> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com >> <mailto:PAYPAL%3Aanthony.minessale@gmail.com> >> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888@conference.freeswitch.org >> <mailto:sip%3A888@conference.freeswitch.org> >> iax:guest@conference.freeswitch.org/888 >> <http://iax:guest@conference.freeswitch.org/888> >> googletalk:conf+888@conference.freeswitch.org >> <mailto:googletalk%3Aconf%2B888@conference.freeswitch.org> >> pstn:213-799-1400 >> >> >> >> >> -- >> ------------------------------------------- >> Apostolos Pantsiopoulos >> Kinetix Tele.com R & D >> email: regs(a)kinetix.gr <mailto:regs@kinetix.gr> >> ------------------------------------------- >> >> ------------------------------------------------------------------------ >> >> >> >> >> _______________________________________________ >> 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 >> >> >> >> >> >> -- >> ------------------------------------------- >> Apostolos Pantsiopoulos >> Kinetix Tele.com R & D >> email: regs(a)kinetix.gr <mailto:regs@kinetix.gr> >> ------------------------------------------- > > > -- > ------------------------------------------- > Apostolos Pantsiopoulos > Kinetix Tele.com R & D > email: regs(a)kinetix.gr > ------------------------------------------- -- ------------------------------------------- Apostolos Pantsiopoulos Kinetix Tele.com R & D email: regs(a)kinetix.gr -------------------------------------------

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