Re: [Freeswitch-users] sip message logging and analysis
I'm using contrib/seven/sip/sip2db.rb 2009/12/18 David Villasmil david.villasmil.w...@gmail.com: i agree with christian, though i would use tshark. you can actually get the fields you want (method and callid) and store them in a dB. then you need to match them with a query. it is simple but Lots of work. look into -e and -E of tshark separate the fields by , have fun! David El 18/12/2009, a las 01:27, Kristian Kielhofner kristian.kielhof...@gmail.com escribió: Frank, Probably the cleanest (albeit non-FreeSWITCH) way to implement this would be to use OpenSIPS/SER/etc between you and the carrier with the siptrace module. But that's probably more work than you want. There's always tcpdump with a decent filter (udp port 5060 and host x.x.x.x) and then something like http://www.badpenguin.co.uk/files/pcap-util2 Both will allow you to search for BYEs and who is sending them. Also keep in mind that they (or you) may just be dropping the RTP without ever sending a BYE. Setting the various RTP timeouts in FreeSWITCH can help with that. You can then look for logs/events (are there any for RTP timeout?) to see who's dropping RTP. On Thu, Dec 17, 2009 at 7:01 PM, Frank @ Impact fr...@impactfax.com wrote: I bit off topic but… Using FS to send calls sip to the LD carrier. Some calls have problems where they drop the call or audio drops or whatever. The carrier’s first response is that we dropped the call. But thi s is a day later after the trouble has been reported. I am looking for guidance on how to log all sip message traffic and then be able to easily retrieve to find a call and look at what sip messages really were being based and by whom. Maybe store them in a database or some other file that might be opened by an analysis tool. Any suggestions on how to log this information and then what tool to use for later analysis? ___ FreeSWITCH-users mailing list 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 -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com ___ FreeSWITCH-users mailing list 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@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] How can I join two freeswitch on two servers?
I couldn't guess what you want, pastbin your full config and logs and give more detail of your story perhaps someone can help you. 2009/12/18 yvonne ding yhding2...@yahoo.ca: param name=username value=1101 param name=password value=1234 param name=proxy value=192.168.129.194:5060 param name=register value=false Hi, If I configure data as following, why FS A 1001 call FS B 1003 failed ? Thank you! FS A: 192.168.129.168, DN=1001 FS B: 192.168.129.194, DN=1003 In FS A add /conf/sip_proifles/external/gwfsa.xml include gateway name=gwfsa /gateway /include 1101 is configured in FS B /conf/directory/default/1101.xml, FS A don't have 1101 number Dan Le wrote: If you want FS server A to be able to call FS server B, you can set up a user account in server B's FS directory configs, and then just treat server B as a normal gateway by adding a gateway definition in server A. That will allow you to route calls to server B from A; to do the reverse, just mirror the configs the other direction. On Mon, Jun 15, 2009 at 9:38 PM, Edmar Cruz darklio...@yahoo.com wrote: I like to connect two freeswitch, call each other, communicate and vice versa. Can you give me an example for that? Thanks -- View this message in context: http://www.nabble.com/How-can-I-join-two-freeswitch-on-two-servers--tp24045824p24045824.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ Freeswitch-users mailing list 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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- View this message in context: http://old.nabble.com/How-can-I-join-two-freeswitch-on-two-servers--tp24045824p26832823.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ FreeSWITCH-users mailing list 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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] fifo caller id
I think if you listen to CUSTOM FIFO::INFO you can get Caller-Caller-ID-Number on event socket. 2009/12/13 Diego Toro dft...@yahoo.com Hello, I want to know how can I get caller id after call is out queue fifo, I read about fifo_caller_consumer_import and fifo_consumer_caller_import variables, but i don't know use it. I appreciate any suggestion Diego Toro http://lacarretade.blogspot.com/ ___ FreeSWITCH-users mailing list 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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Sofia performance
you can use the same ip with different port 2009/12/13, Yehavi Bourvine yehavi.bourv...@gmail.com: Hello, In the WIKI page that talks about Freeswitch performance there is a sentence: *libsofia only handles 1 thread per profile, so if that is your bottle neck use more profiles* How can I enable more than one profile on the same interface? Won't they colide when using the same IP and port? Thanks! __Yehavi: ___ FreeSWITCH-users mailing list 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
Re: [Freeswitch-users] att_xfer origination_cancel not working and b has no chance to talk with c in an A-B-C scenario
Thanks, done. 2009/12/7 Michael Jerris m...@jerris.com: Please report bugs to jira.freeswitch.org. Mike On Dec 6, 2009, at 11:45 PM, Seven Du wrote: Hi, I know there's some chang on att_xfer, and after upgrade(re-bootstrap) to trunk code, no sound after att_xfer. Then I rebuild FS 15807 with a fresh checkout, but still using the old conf/ settings, sound is ok, but there are other problems: A call B, and B att_xfer C 1) origination_cancel_key not working. no even no DTMF log in FS when I press # or any other key, I tried with Zoiper and Snom(on the B leg) 2) when C answers, B immediately hangup, so B has no chance talk to C Could this be a problem? I pasted logs: http://pastebin.freeswitch.org/11417 ___ FreeSWITCH-users mailing list 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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Zombie Records in core db
I also have this problem on a trunk version more than 1000 revisions behind, so I think the best way is to upgrade to trunk and report this again if still have problem. 2009/12/8 DJB djbin...@yahoo.com: We have FreeSWITCH Version 1.0.4 (exported) running at a high volume traffic. I normally check the concurrent calls by looking at the number of sessions from status command. However, the number of concurrent calls in FS is normally higher than it's supposed to be after we ran traffic for about a week. Thus, I routed the traffic away from the FS and found out from show calls that there were so many old calls from previous days. We are running a pass-thru traffic in signaling only. I wonder whether there is a way to have those zombied records clean up automatically. Also, what should I do to prevent this problem? Thank you, Dorn B. ___ FreeSWITCH-users mailing list 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@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] att_xfer origination_cancel not working and b has no chance to talk with c in an A-B-C scenario
Hi, I know there's some chang on att_xfer, and after upgrade(re-bootstrap) to trunk code, no sound after att_xfer. Then I rebuild FS 15807 with a fresh checkout, but still using the old conf/ settings, sound is ok, but there are other problems: A call B, and B att_xfer C 1) origination_cancel_key not working. no even no DTMF log in FS when I press # or any other key, I tried with Zoiper and Snom(on the B leg) 2) when C answers, B immediately hangup, so B has no chance talk to C Could this be a problem? I pasted logs: http://pastebin.freeswitch.org/11417 Thanks. ___ FreeSWITCH-users mailing list 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
Re: [Freeswitch-users] Generate cdrs
2009/12/4 Mouncifbb mounci...@gmail.com: I don't want to use XML cdr it puts each call on individual files so It posts to a http server, and fall back to a xml file if server fails is it possible to include a JavaScript at the end of dialplan to collect info about the session? I think the answer is yes but where would you store the collected info? Thanks On Dec 3, 2009, at 7:02 PM, Seven Du dujinf...@gmail.com wrote: why not try mod_xml_cdr? 2009/12/4 Mouncif Benniane mounci...@gmail.com: is it possible to run a javascript at the end of dialplan to generate cdrs? because (mod_cdr_csv) is giving me hard time as it rotates Master file on machine reboots or shutdown signals. javascript or LUA for preferences? thank you ___ FreeSWITCH-users mailing list 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@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@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] How to run a JS script periodically
Not sure about js, but in lua, you can use luarun to run a long-running script like loop do sth. sleep 5min end and also it can be set to start with freeswitch in lua.conf.xml I guess you can also use jsrun to run js. And, if you run every 5 min, why not use crontab? fs_cli -x jsrun xx.js 2009/12/3 Oscav os...@hotmail.fr: Hi, Someone knows how to run periodically a JS script ?? The purpose is to write to a db some global informations (Global Variables) about FS like every 5 minutes. Thanks. -- View this message in context: http://old.nabble.com/How-to-run-a-JS-script-periodically-tp26625147p26625147.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ FreeSWITCH-users mailing list 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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Generate cdrs
why not try mod_xml_cdr? 2009/12/4 Mouncif Benniane mounci...@gmail.com: is it possible to run a javascript at the end of dialplan to generate cdrs? because (mod_cdr_csv) is giving me hard time as it rotates Master file on machine reboots or shutdown signals. javascript or LUA for preferences? thank you ___ FreeSWITCH-users mailing list 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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Cannot Do this Basic thing
You didn't say the exact error was. was 10.15.0.91 == aaa.bbb.ccc.ddd ? 2009/12/4 Samuel Abekah-Mensah ab...@greatiam.com: Hi Sorry .xm is a typo. I actually shut down the server and restarted. The log says I need to create a domain of aaa.bbb.ccc.ddd (which is the server IP address ) and then put the user in that domain. Isn't the default domain that of the server FS is running on ? 2319.xml is in /usr/local/freeswitch/conf/directory/default/ Thanks for your time Michael Collins wrote: On Thu, Dec 3, 2009 at 9:46 AM, Samuel Abekah-Mensah ab...@greatiam.com mailto:ab...@greatiam.com wrote: I have copied 1001.xml in directory/default to a test user 2319.xm changing or instances of 1001 in the file to 2319. I then went into default.xml in directory folder and in one of the groups just mimicked 1001 details by changing 1001 to 2319. Connecting to FS gives Forbidden message. However 1001 connects without a problem. What have I missed ? Is there a place that just puts things in do this and that and that to create a new user ? Did you execute reloadxml from the fs cli before trying to connect with 2319? Also I'm assuming that 2319.xm is a typo and you actually created 2319.xml in the default/directory subdir. -MC ___ FreeSWITCH-users mailing list 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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Recording with Native File PCMU
http://code.google.com/p/mod-recpld/ It's out-dated. I originally wrote it to record raw G.729 codec on passthrough mode. It worked before and then we abandoned that since We felt G729 cannot deliver good sound particularly on a cross-continent network. The code is written when I don't know much about FS internals, perhaps it's easier to write some mod with indicate no-transcoding in switch_rtp.c. 2009/11/25 Rupa Schomaker r...@rupa.com Yes On Wed, Nov 25, 2009 at 9:44 AM, Matthew Fong mattdf...@gmail.com wrote: so is using session_record with .wav my best option for recording bridged calls? --matt On Wed, Nov 25, 2009 at 7:18 AM, Brian West br...@freeswitch.org wrote: These two options attach media bugs on to the session. Which doesn't work with native files as far as I know. /b On Nov 25, 2009, at 8:53 AM, Matthew Fong wrote: record WORKS!, but uuid_record and session_record do not want to record in native format. do uuid_record and session_record work with native format? or is it not going to be possible to record a bridged call in native format?...maybe because there are two different channels with a bridged call? ___ FreeSWITCH-users mailing list 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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- -Rupa ___ FreeSWITCH-users mailing list 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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] how to enable short recordings
And you may also would like to update the wiki as well if the var is not there. 2009/11/26 kokoska rokoska kokoska.roko...@post.cz Thank you very much, Anthony, for your help! I'm nearly at current trunk (15653) and action application='set' data='RECORD_MIN_SEC=1'/ works great :-) Many thanks once more! Best regards, kokoska.rokoska Anthony Minessale napsal(a): use the variable RECORD_MIN_SEC This was added in revision 15271 so if you are below that I recommend updating to latest trunk. On Wed, Nov 25, 2009 at 10:03 AM, kokoska rokoska kokoska.roko...@post.cz mailto:kokoska.roko...@post.cz wrote: Yes, Brian, I need them :-) They don't contain speech - instead, they contain few computer generated tones and I should store them in max quality for later proccessing (i.e. analysis)... Best regards, kokoska.rokoska Brian West napsal(a): Really you want to keep 1-3 second files around? /b On Nov 25, 2009, at 9:46 AM, kokoska rokoska wrote: Thank you very much, Brian, for your interest! It is standard recording: action application='record_session' data='/path/file.wav' / Best regards, kokoska.rokoska ___ FreeSWITCH-users mailing list FreeSWITCH-users@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@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/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com mailto:msn%3aanthony_miness...@hotmail.commsn%253aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com mailto:paypal%3aanthony.miness...@gmail.compaypal%253aanthony.miness...@gmail.com IRC: irc.freenode.net http://irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org mailto:sip%3a...@conference.freeswitch.orgsip%253a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 http://iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@conference.freeswitch.org mailto:googletalk%3aconf%2b...@conference.freeswitch.orggoogletalk%253aconf%252b...@conference.freeswitch.org pstn:213-799-1400 ___ FreeSWITCH-users mailing list 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@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Recording with Native File PCMU
Yeah, that's why I had to record to two files(readwrite) and need to mix together by using sox. Do you only try to using PCMU to save CPU power matt? As Anthony said, the difference can be ignored. And you also need to take extra effort to make sure transcoding will not happen on a conversation. But it maybe useful for expensive codecs like g729, iLBC, speex etc for recording heavy scenarios. I'd like to take a look if there is a 5k bounty ;) 2009/11/26 Anthony Minessale anthony.miness...@gmail.com The processor power saved is negligible between PCMU and raw PCM and not worth the fuss. If you didn't decode the audio first you would not be able to mix the stream to produce a single file. So if we went to the trouble of making native media bugs to be able to do that you could barely use them so it would not be worth the 5k or more bounty to develop that functionality. On Sun, Nov 22, 2009 at 12:48 PM, Matthew Fong mattdf...@gmail.comwrote: I'm trying to conserve processor power by recording in native file format, PCMU in my case. It works great with the following line session:execute(record, /tmp/my_recording...session:getVariable(read_codec)); however it fails to work with session:execute(record_session, /tmp/my_recording...session:getVariable(read_codec)); or record = api:execute(sched_api, '+1 none uuid_record '..session:getVariable(uuid)..' start /tmp/my_recording.'..session:getVariable(read_codec)); Why is it that it works with record, but not with record_session or uuid_record? Is there something I'm over looking? In the latter two the consul reports 2009-11-22 18:39:04.265284 [INFO] mod_native_file.c:82 Opening File [/tmp/my_recording.PCMU] 8000hz as if it's recording, but /tmp/my_recording.PCMU never shows up. However if I change it to .wav instead of .PCMU it works. Any ideas? --matt ___ FreeSWITCH-users mailing list 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/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@conference.freeswitch.org pstn:213-799-1400 ___ FreeSWITCH-users mailing list 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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] FIFO Orgination_caller_id
And because it's static string for on-hook members, it's hard to set
dynamically. For now, I'm using a callback way - whenever the sip client
answered the call, it fetch the real connected number from a http server.
That's not ideal because not only it add the complexity but also the callee
have no idea what the number is before answer.
The problem for on-hook agent is that it call the agent first, and then get
one customer from the fifo queue, so it is not possible to let the agent
know the real caller-id before answer. Ideas?
2009/11/24 Anthony Minessale anthony.miness...@gmail.com
if you add
{origination_caller_id_name=foo,origination_caller_id_number=123} before the
static entries for the on hook agent it will prevail over the default one.
If you are using 1.0.4, this feature is only available in trunk or one of
the 1.0.5 pre releases.
On Mon, Nov 23, 2009 at 4:49 PM, Adam Ford li...@redbonez.net wrote:
Is there any way to set the origination_caller_id for a FIFO outbound
call to an on-hook agent? I can’t find anything in the wiki about a FIFO or
member variable to set this. It seems to be set to ‘Queue’ by default, and
appears to be hardcoded in the module source. It would be nice to be able
to change per FIFO queue. That way agents that handle multiple companies
can more easily see which queue is calling and answer accordingly.
It is not a big deal, since it does automatically set the
origination_caller_id_number to ‘fifo+fifo name’. However, depending on
the phone, the caller ID number is not always readily shown, and must be
looked for.
Thanks to anyone who has some insight on this,
-Adam
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Re: [Freeswitch-users] FIFO Orgination_caller_id
Yes, that's what we are doing. 2009/11/24 Brian West br...@freeswitch.org You do realize that the whole concept is OLD skewl. You should be popping this info via external resources when the agent is bridged to the caller and the info is there before they are done saying thanks for calling spacely sprockets, this is George how may I help you /b On Nov 23, 2009, at 7:07 PM, Michael Collins wrote: Tony and Brian were discussing this today. They bring up a really good point: do you want to risk having calls remain on hold as they bounce around looking for an agent? This can happen if you pre- determine which caller goes to which agent and the agent doesn't answer. I do understand why this feature matters to many people - it's how old school ACD systems work. However, mod_fifo is more efficient. It's hard to justify decreasing call routing efficiency in order to display the caller's info to the on-hook agent prior to answering. -MC ___ FreeSWITCH-users mailing list 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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Business/holiday hours routing
XML has basic conditioning, but lua rocks. -- Time condition for sales 1 --session:setAutoHangup(false) function do_transfer(extn) --print(extn) session:transfer(extn, XML, sales) end now = os.date(%H:%M) w = tonumber(os.date(%w)) if w = 1 and w =5 then if ( now = 09:00 and now 20:30 ) then do_transfer(sales_fifo_1) elseif ( now = 20:30 and now 22:30 ) then do_transfer(sales_fifo_2) else do_transfer(sales_fifo_cellphone) end else if ( now = 10:00 and now 19:00 ) then do_transfer(sales_fifo_1) elseif (now = 20:00 and now 22:30 ) then do_transfer(sales_fifo_2) else do_transfer(sales_fifo_cellphone) end end 2009/11/24 Adam Ford li...@redbonez.net Is there a standard module for FreeSWITCH out there that people use for routing calls based on business hours and a holiday schedule? Or is everyone just creating their own in the XML dialplan?(which seems pretty simple) I can’t seem to find anything on the wiki, but might just be searching for the wrong thing. I am relatively new at FreeSWITCH and would rather follow what the community has decided is the best practice, instead of trying to reinvent the wheel myself. Thanks for any input, -Adam ___ FreeSWITCH-users mailing list 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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Recording with Native File PCMU
did you try without any .wav or .PCMU? 2009/11/23 Matthew Fong mattdf...@gmail.com I'm trying to conserve processor power by recording in native file format, PCMU in my case. It works great with the following line session:execute(record, /tmp/my_recording...session:getVariable(read_codec)); however it fails to work with session:execute(record_session, /tmp/my_recording...session:getVariable(read_codec)); or record = api:execute(sched_api, '+1 none uuid_record '..session:getVariable(uuid)..' start /tmp/my_recording.'..session:getVariable(read_codec)); Why is it that it works with record, but not with record_session or uuid_record? Is there something I'm over looking? In the latter two the consul reports 2009-11-22 18:39:04.265284 [INFO] mod_native_file.c:82 Opening File [/tmp/my_recording.PCMU] 8000hz as if it's recording, but /tmp/my_recording.PCMU never shows up. However if I change it to .wav instead of .PCMU it works. Any ideas? --matt ___ FreeSWITCH-users mailing list 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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] mod_fifo and multi-tenancy
I once wrote a patch for fifo delete, but didn't submit to jira. If someone think it's useful to merge into trunk, I think I can still find the code, but sure need to test with the current trunk. 2009/11/19 Michael Collins m...@freeswitch.org On Wed, Nov 18, 2009 at 5:09 PM, mayamatakeshi mayamatake...@gmail.comwrote: On Thu, Nov 19, 2009 at 6:36 AM, Michael Collins m...@freeswitch.orgwrote: On Wed, Nov 18, 2009 at 8:32 AM, mayamatakeshi mayamatake...@gmail.comwrote: About mod_fifo, it would be safe to use it in multi-tenancy scenarios where domains are created and deleted all the time and in consequence, fifos are created all the time? I mean, fifos are eventually destroyed by mod_fifo itself. Is this correct? br, takeshi No, FIFOs are not destroyed automatically just because the last member goes away. Tony says that an empty FIFO takes up almost no memory so performance shouldn't be an issue. You can always issue the API command: *fifo reparse del_all* to clean everything out if you feel like things are getting out of hand. Thanks, I have updated the mod_fifo wiki page. FYI, I was doing some other research and I noticed this on the mod_fifo wiki page: *fifo_destroy_after_use*: FreeSWITCH automatically create a new FIFO when the first time use it, and keep in the memory hash. This var tell FreeSWITCH destroy it to save memory. Using for a one time FIFO. So... you do have that option as well. -MC ___ FreeSWITCH-users mailing list 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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Changing User-Agent String
lol: 2009/11/19 Anthony Minessale anthony.miness...@gmail.com maybe you could send them 183 then 4 180's or send them an invite and pretend to deadlock and not send any more sip traffic as a way of identifying yourself On Wed, Nov 18, 2009 at 4:05 PM, Brian West br...@freeswitch.org wrote: Well this is a bit more informal vs the wiki where people take it as fact! :) Plus its a little humpday humor! /b On Nov 18, 2009, at 3:30 PM, Rob Forman wrote: lol! we have to play nice in the wiki but the mailing list is another story. On Nov 18, 2009, at 3:20 PM, Brian West wrote: Sounds like you need to take a baseball bat to their forehead. /b ___ FreeSWITCH-users mailing list 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/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@conference.freeswitch.org pstn:213-799-1400 ___ FreeSWITCH-users mailing list 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@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] prefix Freeswitch-users vs. FreeSWITCH-Users
Would it be better to change the list subject prefix from [Freeswitch-users] to FreeSWITCH-Users? ___ FreeSWITCH-users mailing list 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
Re: [Freeswitch-users] mod_skypiax for OSX?????
2009/11/6 Giovanni Maruzzelli gmar...@celliax.org On Thu, Nov 5, 2009 at 6:57 PM, Seven Du dujinf...@gmail.com wrote: Ciao Giovanni, Do you still plan to merge this? Sorry Seven, I've lost track of this, and now I'm so sick I'm completely un-useful ;). That's OK, we all have a lot of things to do each day. But yes, I would like to do it, if you think it is in a useful state. Can you please create a Jira and attach an svn diff, so in the next days I can merge it? I'd like to create a jira and I think it would be easier if you can directly merge from branch. However the branch is a bit old and it would need some days if you need svn diff based on the current trunk. Thanks. -giovanni 2009/9/5 Giovanni Maruzzelli gmar...@celliax.org Seven, thanks a lot for your efforts. I will merge it in the next days, and I will take care that it will not breaks Windows or Linux. If I find problems I will wait for you having more time in the future. I send you my super best wishes for your personal things to go well and solves in the best of the possible ways. ciao for now, -giovanni Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 On Sat, Sep 5, 2009 at 1:13 PM, Seven Dudujinf...@gmail.com wrote: gm, Thanks a lot you can merge into the mainline. I check into my branch because it's currently not as useful as on Linux and Windows and the solution is not good. But it works and it is a good start that mod_skypiax runs on OSX. Sure it would be easier for people want to test and improve it if it been merged into trunk. I think you can make a diff by svn diff -r 14472:14772 http://svn.freeswitch.org/svn/freeswitch/branches/seven/src/mod/endpoints/mod_skypiax FYI for personal reason I won't have much time put on this in the coming month. Actually the code was done a few weeks ago but i only got a chance to commit it yesterday. Sure that is not to say I cannot do but fixes. But can you please make sure it won't break Linux/ windows build when you merge the code? I haven't have a chance to test all of them yet. -7- On Sep 5, 2009, at 4:49 PM, Giovanni Maruzzelli wrote: Seeeven! I saw the modification you made on the wiki page... You made it, mod_skypiax runs on OSX Let's merge your mods on the mainline, plese ;-))) -giovanni Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 ___ FreeSWITCH-users mailing list 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@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@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 ___ FreeSWITCH-users mailing list 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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Fwd: Many CS_REPORTING state Zombie session
No, I'm Seven and never used hangup hook. you must had though I was Dome. Sorry, I'm not tend to hijack this thread, just though it's the same topic. 2009/11/2 Anthony Minessale anthony.miness...@gmail.com We already concluded its your unacceptabe use of originate in hangup hook right? On Nov 1, 2009 7:45 PM, Seven Du dujinf...@gmail.com wrote: Just suspicious would be possible that happened on sqlite stage? I manually deleted the channels from sqlite and nothing bad happend. just FYI. -- Forwarded message -- From: Seven Du dujinf...@gmail.com Date: Sun, 1 Nov 2009 10:24:32 +0800 Subject: Fwd: [Freeswitch-users] Many CS_REPORTING state Zombie session To: Thank you Anthony. We are on r14696 and no non-standard mods loaded. we even unloaded mod_xml_cdr. But we are heavily using mod_erlang_event both inbound and outbound. I must be very careful if I upgrade to trunk and turn on rwlock debug because it's on production and the the problem not happening that much so would be hard to trace. But I will find time to test and report back. FYI, I also noticed that in some zombile channels, couple of INVITEs sent but never got a response. However, it sent out an ACK at last. ___ FreeSWITCH-users mailing list freeswitch-us...@list... ___ FreeSWITCH-users mailing list 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@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] Fwd: Many CS_REPORTING state Zombie session
Just suspicious would be possible that happened on sqlite stage? I manually deleted the channels from sqlite and nothing bad happend. just FYI. -- Forwarded message -- From: Seven Du dujinf...@gmail.com Date: Sun, 1 Nov 2009 10:24:32 +0800 Subject: Fwd: [Freeswitch-users] Many CS_REPORTING state Zombie session To: Thank you Anthony. We are on r14696 and no non-standard mods loaded. we even unloaded mod_xml_cdr. But we are heavily using mod_erlang_event both inbound and outbound. I must be very careful if I upgrade to trunk and turn on rwlock debug because it's on production and the the problem not happening that much so would be hard to trace. But I will find time to test and report back. FYI, I also noticed that in some zombile channels, couple of INVITEs sent but never got a response. However, it sent out an ACK at last. ___ FreeSWITCH-users mailing list 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
Re: [Freeswitch-users] Many CS_REPORTING state Zombie session
I also got some zombie channels, if someone can help me take a look that's really nice. http://pastebin.freeswitch.org/10912 I only loaded mod_cdr_csv Is it ok to use mod_xml_cdr? Thanks. 2009/11/1 Dome Charoenyost d...@tel.co.th I found bug in fscore_pb $pwd should be $mypwd Now i post already. please help me check. Dome C. 2009/11/1 João Mesquita jmesqu...@freeswitch.org: No, mod_nibblebill definetely needs to be enhanced but it is not the problem and it can be used with high load traffic. The one I am not sure about is odbc_query since it was not developed for that. Do what Rupa said, please. Regards, JM On Sat, Oct 31, 2009 at 3:11 PM, Brian West br...@freeswitch.org wrote: I think once you get the backtrace like rupa said we can see that maybe odbc_query is really hanging or something similar. /b On Oct 31, 2009, at 12:05 PM, Dome Charoenyost wrote: 2009/10/31 Brian West br...@freeswitch.org: You should never do billing inline with the session thread is all I'm saying. ___ FreeSWITCH-users mailing list 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@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@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] uuid_send_dtmf fails (was: conference call)
not sure about this, but did you try send dtmf to uuid 723f3dbb-b87b-4cd4-98fc-698eed7f2bdb other than cced4b9a-b6de-4be1-8c12- 3d18cc6e8454 ? 2009/10/20 Nikita Belov nbe...@abisoft.spb.ru Yeah. I'm using it for starting eavesdrop: SendMsg cced4b9a-b6de-4be1-8c12-3d18cc6e8454 call-command: execute execute-app-name: eavesdrop execute-app-arg: 723f3dbb-b87b-4cd4-98fc-698eed7f2bdb But we are discussing how to switch eavesdrop to allow spy to speak with one of the sides. Do you know any application which can I use for this? Mike said I should use sendevent command to control eavesdrop, but I still confused how can I do it. -Original Message- From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch- users-boun...@lists.freeswitch.org] On Behalf Of Brian West Sent: Monday, October 19, 2009 8:59 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] uuid_send_dtmf fails (was: conference call) Please review this: http://wiki.freeswitch.org/wiki/Mod_event_socket#sendmsg SendMsg uuid call-command: execute execute-app-name: one of the applications execute-app-arg: application data /b On Oct 19, 2009, at 11:42 AM, Nikita Belov wrote: And what event name to use for sendevent command? Sorry for importunity. sendevent ??? Unique-ID: b9d6a35c-ee0c-4203-8d65-ed816e0a9c19 eavesdrop-command: 1 ___ FreeSWITCH-users mailing list 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@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] freeswitch french community
There are some plugins for mediawiki to support multilanguage, either inline or on a separate page. However, both have Pros and Cons. e.g. multi-language support this with a separate page some_page #default to en some_page/fr Franch some_page/zh Chinese Pros: clear, available languages can be listed in Other languages for this page Cons: hard to keep all languages in sync other plugins can do inline translation: @en|hello @fr|Ca Va @zh|你好 Pros: untranslated items remains in default lang, no sync headache Cons: It breaks a page into sentences or items, also if we have more than 10 languages the source code maybe really a mess. 2009/10/19 Diego Viola diego.vi...@gmail.com The other way we could do is set up a fr.freeswitch.org with mediawiki on it and you start to translate from the English wiki. But I think inline translation would be better for this. Just my two cents. Diego On Sun, Oct 18, 2009 at 11:13 PM, Diego Viola diego.vi...@gmail.comwrote: Hi Meftah, I'm more than happy to help with the documentation and anything else related to FreeSWITCH. As you mention the documentation and translation, there are some things that I would really like to improve in the documentation. I propose that we start a new documentation from the ground up, we really need a better wiki engine that supports inline translation and a better search engine. I'm open to do all the hard work like porting the entire documentation to it. We could put the new wiki on a sub-domain like: newdocs.freeswitch.organd I do all the work there, without affecting others. That way we could have inline documentation like this: text:enstatus - show status information/text text:esstatus - mostrar informacion de estado/text It's up to others to decide if this is really worth or not. I'm open to help and to give my time to improve things like documentation, etc :). Diego On Sun, Oct 18, 2009 at 8:08 PM, Meftah Tayeb tayeb.mef...@gmail.comwrote: hello folk we are starting a new freeswitch community for french users the newly created Irc channel is #freeswitch-fr freeswitch staf, please can we register this channel officialy to the freeswitch freenode group? also can we build a french documentation and host it somehere? in the wiki? my long time friend, diegoviola i going to help about the freeswitch wiki markup we are waiting for french users to join, and welcome thank toAnthony Minessale for accepting the community welcome! __ Information from ESET NOD32 Antivirus, version of virus signature database 4520 (20091018) __ The message was checked by ESET NOD32 Antivirus. http://www.eset.com ___ FreeSWITCH-users mailing list 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@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] mod_opal - call charged before H.225 connect
that will make life easier. 2009/10/13 Brian West br...@freeswitch.org Does anyone see a problem with hosting mod_h323 in our SVN? I would like to centralize everything we can to reuse our issue tracking resources and not fragment the community if possible. /b On Oct 12, 2009, at 2:43 PM, Tihomir Culjaga wrote: hi, finally i compiled it right ... had a stupid issue with ekiga and wrong ptlib in place... anyhow, i loaded the module and will continue the tests tomorrow ...first thing i arrive in my office :P ___ FreeSWITCH-users mailing list 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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Re corded file as voicemail.
http://jira.freeswitch.org/browse/MODCODEC-15 Is it ok I assigned to you ? Thanks. 2009/10/12 Brian West br...@freeswitch.org It was possible but we have a regression in the code that isn't letting that happen right now... hence the reason i said Open a jira so we could fix it. IS THAT not clear? /b On Oct 11, 2009, at 10:46 PM, Nagalenoj wrote: Whats the conclusion.?! ___ FreeSWITCH-users mailing list 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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] mod_sofia.c registered calls how to know
try open YOUR_FreeSWITCH_INSTALL_DIR/db/*.db, you need sqlite3 to open them. not sure how to do that on windows, but on linux: # sqlite3 xx.db sqlite select * from sip_registration; 2009/10/12 srinivasula reddy srinivas.ksvre...@gmail.com Hi Mike, Thanks for your valuable reply, when i install freeswitch1.0.2 in my machine(Windows xp operation system) i dont have any databasae installed in my system, then from sqllite will come into picture, and how can i see the registered users data from sqllite. Thanks Srinivas On Mon, Oct 12, 2009 at 3:04 AM, Michael Jerris m...@jerris.com wrote: On Oct 7, 2009, at 10:48 AM, srinivasula reddy wrote: Hi can any please tell me where registered calls are stored, so when incoming call came to mod_sofia.c how it will check it is registered or not?\\ Calls are not registered and calls have nothing to do with registration. Users are registered so that you may send calls to them. Registration data is stored either in a sqlite database, or optionally if you setup odbc, in another database of your choice. If you try to send a call to an unregistered user in the dialplan using the proper syntax to send calls to registered users (see the wiki for more details), and that user is not registered, the bridge app will fail, optionally letting you continue on in the dialplan based on variables such as continue_on_fail and hangup_after_bridge. You can use the sofia_contact function to see if there is anyone registered to a specific user. Mike ___ FreeSWITCH-users mailing list 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 -- Srinivasula Reddy K ___ FreeSWITCH-users mailing list 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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Re corded file as voicemail.
I won't try until I need that, but I believe it works. Thanks Brian. 2009/10/13 Brian West br...@freeswitch.org Fixed... svn up. /b On Oct 12, 2009, at 1:15 AM, Seven Du wrote: http://jira.freeswitch.org/browse/MODCODEC-15 Is it ok I assigned to you ? Thanks. ___ FreeSWITCH-users mailing list 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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Re corded file as voicemail.
It was a problem and has been fixed in the last trunk. Just update to the latest code should be ok. btw, the developers using jira to track bugs, so feel free to report one (as you see http://jira.freeswitch.org/browse/FSCORE-463) if you think it's a bug next time. 2009/10/12 Nagalenoj nagale...@gmail.com Whats the conclusion.?! Brian West-3 wrote: Please open a jira please this did work but a recent change in switch_core_codec caused this to appear I usually test this regularly but haven't run thru a full run of tests lately. /b ___ FreeSWITCH-users mailing list 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
Re: [Freeswitch-users] Re corded file as voicemail.
:30.349961 [DEBUG] switch_core_session.c:932 Send signal sofia/eqenglish/seven1240 [BREAK] 2009-10-12 01:41:30.349961 [DEBUG] switch_core_state_machine.c:503 (sofia/eqenglish/seven1240) State CONSUME_MEDIA going to sleep 2009-10-12 01:41:30.349961 [DEBUG] switch_core_state_machine.c:398 (sofia/eqenglish/seven1240) Running State Change CS_EXECUTE 2009-10-12 01:41:30.349961 [DEBUG] switch_core_state_machine.c:491 (sofia/eqenglish/seven1240) State EXECUTE 2009-10-12 01:41:30.349961 [DEBUG] mod_sofia.c:173 sofia/eqenglish/seven1240 SOFIA EXECUTE 2009-10-12 01:41:30.349961 [DEBUG] switch_core_state_machine.c:151 sofia/eqenglish/seven1240 Standard EXECUTE EXECUTE sofia/eqenglish/seven1240 playback(/tmp/a.wav) 2009-10-12 01:41:30.349961 [WARNING] switch_core_file.c:133 File has 2 channels, muxing to mono will occur. 2009-10-12 01:41:30.349961 [ERR] switch_core_codec.c:431 Stereo is currently unsupported. please downsample audio source to mono. 2009-10-12 01:41:30.349961 [DEBUG] switch_ivr_play_say.c:1113 Raw Codec Activation Failed l...@8000hz 2 channels 20ms 2009-10-12 01:41:30.349961 [NOTICE] switch_core_state_machine.c:179 Hangup sofia/eqenglish/seven1240 [CS_EXECUTE] [NORMAL_CLEARING] 2009-10-12 01:41:30.349961 [DEBUG] switch_channel.c:1715 Send signal sofia/eqenglish/seven1240 [KILL] 2009-10-12 01:41:30.349961 [DEBUG] switch_core_session.c:932 Send signal sofia/eqenglish/seven1240 [BREAK] 2009-10-12 01:41:30.349961 [DEBUG] switch_core_state_machine.c:491 (sofia/eqenglish/seven1240) State EXECUTE going to sleep 2009-10-12 01:41:30.349961 [DEBUG] switch_core_state_machine.c:398 (sofia/eqenglish/seven1240) Running State Change CS_HANGUP 2009-10-12 01:41:30.349961 [DEBUG] switch_core_state_machine.c:434 (sofia/eqenglish/seven1240) State HANGUP 2009-10-12 01:41:30.349961 [DEBUG] mod_sofia.c:338 Channel sofia/eqenglish/seven1240 hanging up, cause: NORMAL_CLEARING 2009-10-12 01:41:30.349961 [DEBUG] mod_sofia.c:376 Sending BYE to sofia/eqenglish/seven1240 2009-10-12 01:41:30.349961 [DEBUG] switch_core_state_machine.c:46 sofia/eqenglish/seven1240 Standard HANGUP, cause: NORMAL_CLEARING 2009-10-12 01:41:30.349961 [DEBUG] switch_core_state_machine.c:434 (sofia/eqenglish/seven1240) State HANGUP going to sleep 2009-10-12 01:41:30.349961 [DEBUG] switch_core_state_machine.c:476 (sofia/eqenglish/seven1240) State Change CS_HANGUP - CS_REPORTING 2009-10-12 01:41:30.349961 [DEBUG] switch_core_session.c:932 Send signal sofia/eqenglish/seven1240 [BREAK] 2009-10-12 01:41:30.349961 [DEBUG] switch_core_state_machine.c:398 (sofia/eqenglish/seven1240) Running State Change CS_REPORTING 2009-10-12 01:41:30.349961 [DEBUG] switch_core_state_machine.c:613 (sofia/eqenglish/seven1240) State REPORTING 2009-10-12 01:41:30.349961 [DEBUG] switch_core_state_machine.c:53 sofia/eqenglish/seven1240 Standard REPORTING, cause: NORMAL_CLEARING 2009-10-12 01:41:30.349961 [DEBUG] switch_core_state_machine.c:613 (sofia/eqenglish/seven1240) State REPORTING going to sleep 2009-10-12 01:41:30.349961 [DEBUG] switch_core_state_machine.c:411 (sofia/eqenglish/seven1240) State Change CS_REPORTING - CS_DESTROY 2009-10-12 01:41:30.349961 [DEBUG] switch_core_session.c:932 Send signal sofia/eqenglish/seven1240 [BREAK] 2009-10-12 01:41:30.349961 [DEBUG] switch_core_session.c:1068 Session 2178 (sofia/eqenglish/seven1240) Locked, Waiting on external entities 2009-10-12 01:41:30.349961 [NOTICE] switch_core_session.c:1086 Session 2178 (sofia/eqenglish/seven1240) Ended 2009-10-12 01:41:30.349961 [NOTICE] switch_core_session.c:1088 Close Channel sofia/eqenglish/seven1240 [CS_DESTROY] 2009-10-12 01:41:30.349961 [DEBUG] switch_core_state_machine.c:556 (sofia/eqenglish/seven1240) Running State Change CS_DESTROY 2009-10-12 01:41:30.349961 [DEBUG] switch_core_state_machine.c:565 (sofia/eqenglish/seven1240) State DESTROY 2009-10-12 01:41:30.349961 [DEBUG] mod_sofia.c:255 sofia/eqenglish/seven1240 SOFIA DESTROY 2009-10-12 01:41:30.349961 [DEBUG] switch_core_state_machine.c:60 sofia/eqenglish/seven1240 Standard DESTROY 2009-10-12 01:41:30.349961 [DEBUG] switch_core_state_machine.c:565 (sofia/eqenglish/seven1240) State DESTROY going to sleep 2009-10-12 01:41:30.349961 [NOTICE] switch_cpp.cpp:1130 ==DebugVar== gateway_name: skype 2009/10/12 Brian West br...@freeswitch.org FreeSWITCH can play back stereo files it'll just mux them down to mono before playing... can you elaborate on the error you're getting? /b On Oct 10, 2009, at 12:40 AM, Seven Du wrote: Yes, it's discussed before. http://wiki.freeswitch.org/wiki/Channel_Variables#RECORD_STEREO set that var to false before you record. ___ FreeSWITCH-users mailing list 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
Re: [Freeswitch-users] Re corded file as voicemail.
I set to true because brian said it can play stereo files but no lucky for
me.
2009/10/12 Jason White ja...@jasonjgw.net
Seven Du dujinf...@gmail.com wrote:
originate {ignore_early_media=true,RECORD_STEREO=true}sofia/gateway/xx/xx
bridge(sofia/gateway/yy/yy)
Shouldn't that be record_stereo=false for mono recording?
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Re: [Freeswitch-users] Difference between park and valet_park
search this list, just has been discussed. 2009/10/10 velusamy velu velu.techni...@gmail.com Dear All, Could you please any one explain the difference between parking and valet parking? ___ FreeSWITCH-users mailing list 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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Re corded file as voicemail.
Yes, it's discussed before. http://wiki.freeswitch.org/wiki/Channel_Variables#RECORD_STEREO set that var to false before you record. 2009/10/10 Jason White ja...@jasonjgw.net Nagalenoj nagale...@gmail.com wrote: No, When I do voicemail_inject and check through voicemail, it is not playing the file instead reporting the following error. 'Stereo is currently not supported, please downsample to mono.' This has been discussed on the list before. I think the solution was to use sox to convert the files to mono. You might need a script to do this before injecting them into the voicemail. ___ FreeSWITCH-users mailing list 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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Mobile Phone As GSM Gateway....
maybe you can check this: http://www.gsmopen.org/ 2009/10/6 Moiz Chinoy moizchi...@gmail.com Hi, Is it possible to connect a mobile phone (GSM phone) to Freeswitch and use this as a GSM gateway? -- Regards, Moiz Chinoy. ___ FreeSWITCH-users mailing list 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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Dynamic updation of groups in default.xml
change the xml and execute reloadxml in FS console or fs_cli or you can check mod_xml_curl 2009/10/6 srinivasula reddy srinivas.ksvre...@gmail.com Hi, Can any one tell me how to add users dynamically to groups in default.xml, with out restart the freeswitch. Thanks Srinivasula Reddy K ___ FreeSWITCH-users mailing list 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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] wav files compression
FS support recording to mp3 directly through mod_shout but you might not want to use that for performance reason. You can use lame to convert .wav to .mp3 regularly( by crontab if you on linux) or immediately after record(by using iwatch, or listening to event socket to see when the record is done ). 2009/10/3 Keith Wood keith.wood2...@gmail.com I am working on an implementation for managing thousands of IVR within an organization. Right now, I am storing all audio files in wav format, but it quickly become unmanagable because the size of these wav files ( 8 bits mono ) quickly consuming a lot of the disk space. Is there anyway I can store those audio files and still have high quality audio for IVR? I know mp3 is smaller but freeswitch does not support it. any ideas? keith ___ FreeSWITCH-users mailing list 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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] confuse of regex command with |
Thanks. But I think it would be nicer if the regex looks the same as in dialplan. can we add a optional separator arg on this case? regex data|pattern [seperator] regex 10:09|10 : 2009/9/29 Brian West br...@freeswitch.org Yep escape it. /b On Sep 28, 2009, at 10:47 AM, Michael Collins wrote: regex 10|09\|10 ___ FreeSWITCH-users mailing list 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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Sofia regiter to provider with multiple account
I would write a simple lua script to do a round robin hunt. 2009/9/27 Dome Charoenyost d...@tel.co.th 2009/9/26 Brian West br...@freeswitch.org: Why would they require you to have 50 accounts? Doesn't seem sane to me. They provide for pc to phone user and i want to use for my corp. :) /b On Sep 25, 2009, at 9:48 PM, Dome Charoenyost wrote: 2009/9/26 Michael Jerris m...@jerris.com: On Sep 25, 2009, at 10:04 PM, Dome Charoenyost wrote: Dear All, How to config freeswitch for support this case ? 1. FS register to provider about 50 user account. (Each account can't support multiple call in same time) Sofia gateways 2. FS Check account not inuse before call out. mod_limit 3. User account should be round-robin what does this mean? Each user account have own balance (in provider). So i want to use all user ___ FreeSWITCH-users mailing list 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@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Mod_vmd and Session Locking.
what's your rev? I think rev14494 might related to you. 2009/9/27 Vinuth Madinur vinuth.madi...@gmail.com Hello All, I'm trying to do a simple dialer, where I am: 1. Initiating mod_vmd on channel answer. 2. Staying quiet until there is a beep. 3. Leave a message on beep. 4. Hangup. (in my scenario it's guaranteed to hit voicemail.) But the session isn't exiting properly after hangup, with last statement being: switch_core_session.c:1068 Session 1 (sofia/external/1002) Locked, Waiting on external entities. It doesn't go to CS_DONE state. I see [CRIT] level logs when shutting down freeswitch waiting for all these sessions to complete. But all these sessions are getting stuck while attempting to get a write lock on session object. I tried this with both 1.0.3 and 1.0.4 and the behavior is similar. Can someone please help? Or point me toward fixing it? Thanks, Gubbi. ___ FreeSWITCH-users mailing list 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@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] confuse of regex command with |
Hi, is this a bug? freeswi...@internal regex 10|09|10 false freeswi...@internal regex 10|10 true freeswi...@internal regex 10|(09|10) false freeswi...@internal 2009-09-27 11:47:00.815355 [ERR] switch_regex.c:101 COMPILE ERROR: 4 [missing )][(09] the first one should be true? ___ FreeSWITCH-users mailing list 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
Re: [Freeswitch-users] Bind to more than one ethernet interface
sorry when I said on profile I want to say one profile 2009/9/24 Seven Du dujinf...@gmail.com It not possible to use 0.0.0.0 for on profile. however, you can create more sip profiles for each of your interfaces. Search freeswitch-users archievs then you will find similar topics. 2009/9/24 Yehavi Bourvine yehavi.bourv...@gmail.com Hello, I am trying to run FreeSwitch on a machine which has more than one interface, all of them should be used for SIP. The FreeSwitch binds only to the first one. I tried setting bind_server_ip to either auto or 0.0.0.0 but it doesn't help. Any idea what to do? Thanks! _Yehavi: ___ FreeSWITCH-users mailing list 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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Bind to more than one ethernet interface
It not possible to use 0.0.0.0 for on profile. however, you can create more sip profiles for each of your interfaces. Search freeswitch-users archievs then you will find similar topics. 2009/9/24 Yehavi Bourvine yehavi.bourv...@gmail.com Hello, I am trying to run FreeSwitch on a machine which has more than one interface, all of them should be used for SIP. The FreeSwitch binds only to the first one. I tried setting bind_server_ip to either auto or 0.0.0.0 but it doesn't help. Any idea what to do? Thanks! _Yehavi: ___ FreeSWITCH-users mailing list 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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] session record does not for very short calls
I think the file was there but deleted by FreeSWITCH if it thinks it was too short (like 3 seconds?). If I'm not wrong, someone requested this feature becuase FreeSWITCH left too many small recordings. On Sep 17, 2009, at 1:27 AM, João Mesquita wrote: I think you need to upgrade your version before we even take a look at that... You are so far behind trunk right now and lots of things have been changed since then. Not sure if this would solve your problem but not a lot of ppl will look at your problem when you run this version. jmesquita On Wed, Sep 16, 2009 at 2:00 PM, Frank @ Impact fr...@impactfax.com wrote: FreeSWITCH Version 1.0.trunk (12790M) I have this in my DP action application=set data=RECORD_ANSWER_REQ=true/ action application=set data=RECORD_STEREO=true/ action application=record_session data=/mnt/rd/ file.wav/ works fine as long as the call is long enough. But if the call is only, say, 3-4 seconds long (or something very short like that), then the wav file is never created with the audio in it. Is there a work around for this? ___ FreeSWITCH-users mailing list 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@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] CALL FOR VOLUNTEERS: Assisting With FreeSWITCH Subprojects
Hi MC, Months ago we had tried the multi-language plugin on MediaWiki, I know you are still planning to do this but I just want how far it goes. Count me in when you are short of hand. On Sep 15, 2009, at 11:25 PM, Michael Collins wrote: Demuel, Thanks for the input. Yes, we want to avoid chaos. I will work to keep everyone organized. I would welcome a mass of people all trying to do different things because the challenge would simply be to keep them organized. Right now the challenge is in recruiting people to stay with the not-so-glorious aspects of the project, namely documentation, maintenance, and janitorial style subprojects. Please keep the comments and suggestions coming! Thanks, MC On Tue, Sep 15, 2009 at 12:38 AM, dem...@thephinix.org wrote: Hi Michael / FS enthusiast, In my opinion, it will be an undeniably perfect chaos if everyone wants to do everything. One way I can suggest out of this is the following: - sets of persons that does all the stuff on the FS core. I'm not sure if this one will work out but I think these are the only persons who has commit access to the source code and will also be willing to accept modifications, enhancements, etc. - individuals can ask if they can be the maintainer/tester for a particular module. Bug fixing, sending modifications and enhancements will still be subject to the approval of anthm. - an array of persons that will take ownership on what FS can do and provide realiable working examples for it. Currently, it takes amount of pain for a newbie on how to configure SIP because the examples are too confusing and sometimes there is not certainty if this will work on what release of FS or not. - any individual that has time and interest can start owning the porting of FS to other operating system. Like in my case, I am working on making FS ported to FreeBSD ports but I don't know if somebody did it or if he is currently doing it, as to what stage he is in? - there should be a release engineering team. In my opinion, a stable and current release will be much more sane. As with any other successful open source project, we won't be starting to say I want to contribute this and start working on that. We should indicate that you take ownership of this and any comments should be forwarded right unto you. Again, this is just my opinion. You can take some of it or leave it using sudo rm -rf blah/* . There and back again, Demuel I. Bendano a.k.a engrxyz Hi Michael, You can count with me for anything else, like documentation, coding/scripting, or any other FreeSWITCH related stuff. Regards, Diego 2009/9/14 João Mesquita jmesqu...@freeswitch.org You can assign two things to me. 1. libesl code documentation (partially done and Doxygened - needs cleaning) 2. Bug marshal. I am setting up the proper lab environment here to be able to test most stuff. Count me in for any questions I can answer and I am _always_ on IRC jmesquita On Mon, Sep 14, 2009 at 1:33 PM, Michael Collins m...@freeswitch.org wrote: Hello FreeSWITCHers! We are looking for people who are in a position to help out with various subprojects that will help FreeSWITCH to keep growing. We need people to help out in these basic areas: Bug marshals (people who watch JIRA and test bug reports, patches, etc.) Documentation maintainers (people who update the wiki when new stuff comes out, also those familiar with mediawiki administration) Documentation authors (people who write new docs, how-to's, tutorials, examples, etc.) Package maintainers (people who manage Debian debs, RPMs, etc.) Additionally, we are always looking for more folks to assist with answering questions on IRC and the mailing list. It is definitely nice to have people who've gone through the pains of switching to FreeSWITCH (or learning it from scratch) who can assist the steady stream of new users. If you want to help and aren't sure where to go from here then please at least do the following: #1 - Join #freeswitch on irc.freenode.net and hang out as much as possible #2 - Check the recent changes link on wiki.freeswitch.org each day #3 - Join the Friday public conference call and listen in These three things, in addition to the mailing list, will keep you well in tune with the FreeSWITCH community and what's happening. Next, make a note of the parts of FS that you use frequently, know a lot about, or are particularly passionate about. Those are the items we'd love to have you help us with. For example: if you use mod_xml_curl frequently and have been through the set up process then you're a prime candidate to help answer questions, refine the mod_xml_curl wiki documentation, write up a tutorial, contribute a working example of a web server database schema, etc. If you are
Re: [Freeswitch-users] Recording Only 1 Leg of a Call
As a work around, record to stereo, and use sox to split channes ?
On Sep 9, 2009, at 12:44 AM, Anthony Minessale wrote:
that would have to be filed as a feature request as we do not
currently have a way to do that.
On Mon, Sep 7, 2009 at 11:50 PM, Matthew Fong mattdf...@gmail.com
wrote:
I want to record without the telephone user's interaction.
I think uuid_record should have the option to only record the audio
of the uuid channel that is being specified, and as a secondary
option combine the audio of the b leg (since uuid_record only
specifies one uuid anyway--this seems logical).
Anyway, just my wish list :)
--matt
http://www.hellohunter.com
voice broadcasting hosted dialer
On Tue, Sep 8, 2009 at 2:12 AM, Milena testeado...@gmail.com wrote:
Hello,
What about this?:
!-- bind_meta_app can have these args key [a|b|ab] [a|b|o|s]
app --
action application='bind_meta_app' data='2 a s record_session::$$
{base_dir}/recordings/${strftime(%Y-%m-%d_%H-%M-%S)}.$
{caller_id_number}.wav'/
the person would have to press *2 during the call to start the
recording.
2009/9/7 Matthew Fong mattdf...@gmail.com
Whats the best way to record only one leg of a call?
uuid_record records both channels
session_record does the same (but has a stereo option)
is there any way to record only an a-leg of the call? Thanks so much.
--matt
http://www.hellohunter.com
hosted dialer voice broadcasting
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Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/
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Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm
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FreeSWITCH Developer Conference
sip:8...@conference.freeswitch.org
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Re: [Freeswitch-users] mod_skypiax for OSX?????
I'm not sure this and don't have time to debug. But last time I tried incoming calls worked by setting skype to Auto-answer calls. Can you try that? On Sep 6, 2009, at 3:19 PM, Ivan C Myrvold wrote: I have got outgoing call to Skype to work, and the audio quality is excellent. But I also have problems with incoming call, looks like it doesn't get bridged by FreeSWITCH. I put sofia on debug, but couldn't see any Sofia messages at all. Only message I saw in the console was this: 2009-09-05 20:44:39.720943 [WARNING] skypiax_protocol.c:375 rev 14772[0x0|37 ][WARNINGA 375 ][interface1][-1, 0, 0] skype_call: 332, STATUS: RINGINGCALL is not recognized But I am thrilled that Seven have got skypiax to work this far, and am confident that we will have it worked both incoming and outgoing soon. Ivan Den 6. sep.. 2009 kl. 01:25 skrev Tapan Parikh: Yes, thanks a lot! Im still having a bit of trouble getting it working though. When skypiax_proxy starts, I get the following errors: 2009-09-05 15:39:40.631 skypiax_proxy[77842:10b] Failed to init theDOProxy 2009-09-05 15:39:42.139 skypiax_proxy[77842:10b] Failed to init theDOProxy Then, when an incoming call comes in, I see the messages: Message received CALL 683 STATUS RINGING Sent 23 bytes to FreeSWITCH Sending to skype: GET CALL 683 PARTNER_HANDLE Send to skype: GET CALL 683 PARTNER_HANDLE Message received CALL 683 CONF_ID 0 Sent 18 bytes to FreeSWITCH Message received CALL 683 FAILUREREASON 14 Sent 25 bytes to FreeSWITCH Message received CALL 683 STATUS MISSED Sent 22 bytes to FreeSWITCH Message received CALL 683 SEEN FALSE Sent 19 bytes to FreeSWITCH But the call is not actually picked up by the proxy. Any ideas? Im on Mac OS X 10.5, w/ freeswitch from SVN. Thanks! On Sat, Sep 5, 2009 at 4:41 AM, Giovanni Maruzzelli gmar...@celliax.org wrote: Seven, thanks a lot for your efforts. I will merge it in the next days, and I will take care that it will not breaks Windows or Linux. If I find problems I will wait for you having more time in the future. I send you my super best wishes for your personal things to go well and solves in the best of the possible ways. ciao for now, -giovanni Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 On Sat, Sep 5, 2009 at 1:13 PM, Seven Dudujinf...@gmail.com wrote: gm, Thanks a lot you can merge into the mainline. I check into my branch because it's currently not as useful as on Linux and Windows and the solution is not good. But it works and it is a good start that mod_skypiax runs on OSX. Sure it would be easier for people want to test and improve it if it been merged into trunk. I think you can make a diff by svn diff -r 14472:14772 http://svn.freeswitch.org/svn/freeswitch/branches/seven/src/mod/endpoints/mod_skypiax FYI for personal reason I won't have much time put on this in the coming month. Actually the code was done a few weeks ago but i only got a chance to commit it yesterday. Sure that is not to say I cannot do but fixes. But can you please make sure it won't break Linux/ windows build when you merge the code? I haven't have a chance to test all of them yet. -7- On Sep 5, 2009, at 4:49 PM, Giovanni Maruzzelli wrote: Seeeven! I saw the modification you made on the wiki page... You made it, mod_skypiax runs on OSX Let's merge your mods on the mainline, plese ;-))) -giovanni Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 ___ FreeSWITCH-users mailing list 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@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@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@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
Re: [Freeswitch-users] mod_skypiax for OSX?????
On Sep 6, 2009, at 7:25 AM, Tapan Parikh wrote: Yes, thanks a lot! Im still having a bit of trouble getting it working though. When skypiax_proxy starts, I get the following errors: 2009-09-05 15:39:40.631 skypiax_proxy[77842:10b] Failed to init theDOProxy 2009-09-05 15:39:42.139 skypiax_proxy[77842:10b] Failed to init theDOProxy I don't have that problem. google found this: https://developer.skype.com/jira/browse/SPA-418 Then, when an incoming call comes in, I see the messages: Please try set Auto-Answer on Skype Preference and retry. Message received CALL 683 STATUS RINGING Sent 23 bytes to FreeSWITCH Sending to skype: GET CALL 683 PARTNER_HANDLE Send to skype: GET CALL 683 PARTNER_HANDLE Message received CALL 683 CONF_ID 0 Sent 18 bytes to FreeSWITCH Message received CALL 683 FAILUREREASON 14 Sent 25 bytes to FreeSWITCH Message received CALL 683 STATUS MISSED Sent 22 bytes to FreeSWITCH Message received CALL 683 SEEN FALSE Sent 19 bytes to FreeSWITCH But the call is not actually picked up by the proxy. Any ideas? Im on Mac OS X 10.5, w/ freeswitch from SVN. Thanks! On Sat, Sep 5, 2009 at 4:41 AM, Giovanni Maruzzelli gmar...@celliax.org wrote: Seven, thanks a lot for your efforts. I will merge it in the next days, and I will take care that it will not breaks Windows or Linux. If I find problems I will wait for you having more time in the future. I send you my super best wishes for your personal things to go well and solves in the best of the possible ways. ciao for now, -giovanni Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 On Sat, Sep 5, 2009 at 1:13 PM, Seven Dudujinf...@gmail.com wrote: gm, Thanks a lot you can merge into the mainline. I check into my branch because it's currently not as useful as on Linux and Windows and the solution is not good. But it works and it is a good start that mod_skypiax runs on OSX. Sure it would be easier for people want to test and improve it if it been merged into trunk. I think you can make a diff by svn diff -r 14472:14772 http://svn.freeswitch.org/svn/freeswitch/branches/seven/src/mod/endpoints/mod_skypiax FYI for personal reason I won't have much time put on this in the coming month. Actually the code was done a few weeks ago but i only got a chance to commit it yesterday. Sure that is not to say I cannot do but fixes. But can you please make sure it won't break Linux/ windows build when you merge the code? I haven't have a chance to test all of them yet. -7- On Sep 5, 2009, at 4:49 PM, Giovanni Maruzzelli wrote: Seeeven! I saw the modification you made on the wiki page... You made it, mod_skypiax runs on OSX Let's merge your mods on the mainline, plese ;-))) -giovanni Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 ___ FreeSWITCH-users mailing list 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@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@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@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] mod_skypiax for OSX?????
gm, Thanks a lot you can merge into the mainline. I check into my branch because it's currently not as useful as on Linux and Windows and the solution is not good. But it works and it is a good start that mod_skypiax runs on OSX. Sure it would be easier for people want to test and improve it if it been merged into trunk. I think you can make a diff by svn diff -r 14472:14772 http://svn.freeswitch.org/svn/freeswitch/branches/seven/src/mod/endpoints/mod_skypiax FYI for personal reason I won't have much time put on this in the coming month. Actually the code was done a few weeks ago but i only got a chance to commit it yesterday. Sure that is not to say I cannot do but fixes. But can you please make sure it won't break Linux/ windows build when you merge the code? I haven't have a chance to test all of them yet. -7- On Sep 5, 2009, at 4:49 PM, Giovanni Maruzzelli wrote: Seeeven! I saw the modification you made on the wiki page... You made it, mod_skypiax runs on OSX Let's merge your mods on the mainline, plese ;-))) -giovanni Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 ___ FreeSWITCH-users mailing list 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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] restart when convenient
freeswi...@foosball fsctl -USAGE: [send_sighup|hupall|pause|resume|shutdown [cancel|elegant|asap| restart]|sps|sync_clock|reclaim_mem|max_sessions|max_dtmf_duration [num]|loglevel [level]] On Sep 4, 2009, at 3:35 PM, Anatoliy Kounitskiy wrote: After some testing (fs_cli -x 'fsctl shutdown restart') I'm seeing that all active calls are dropped and the freeswitch is restarted On Fri, Sep 4, 2009 at 10:14 AM, Michael Collinsm...@freeswitch.org wrote: 2009/9/3 Christian Löschenkohl christian.loeschenk...@xpirio.com hello i'm looking for a possibility to restart freeswitch like it is possible with asterisk. for me i tried to created a script that looks for open channels and if no channel is open it restarts freeswitch with the init script (not the most efficient way). i think i would be great if we would have a buildin function for this, i think such command would help with maintenance and not only for me. br Like JM said, the fsctl API can help. If you're in Linux you can do a shell script with a command like ths: fs_cli -x 'fsctl shutdown restart' -MC ___ FreeSWITCH-users mailing list 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 -- Anatoliy Kounitskiy - E-mail: anato...@kounitskiy.com Mobile: +359898913540 ___ FreeSWITCH-users mailing list 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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] select batchfile after call
hangup hook api? system or system_bg ? not sure On Sep 3, 2009, at 9:08 PM, NOx-WHV wrote: Hi, does anybody have a tip how to start a batchfile after hanging up. After ext. 1000 calls 1001 and hang up, i need a request to call: /../../FS/batchfile 1000 if 1001 calls 1000 i need: /../../FS/batchfile 1001 and so on... Thanks for help -- View this message in context: http://www.nabble.com/select-batchfile-after-call-tp25275633p25275633.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ FreeSWITCH-users mailing list 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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] wav2mp3 conversion inside of spidermonkey
I run into this problem before. Don't remember the exact error but might be segfault of lame runing in freeswitch-lua. If you use Linux you would like to try iwatch. It's a perl program watching your file system and can execute the lame command as soon as it got the CLOSE_WRITE(or other) filesystem event. On Aug 26, 2009, at 4:44 PM, Michael Jerris wrote: Running out of stack space? The stack space we run freeswitch in is fairly small. Programs launched from the freeswitch process inherit this. Mike On Aug 26, 2009, at 4:28 AM, Alberto Escudero-Pascual (lists) wrote: I ran strace from freeswitch and from the command line. lame segfaults when run from system FS. The only obvious different i see is in the execve() /* XX vars */ apart from the final Segfault From execve(/usr/local/freeswitch/bin/lame, [/usr/local/freeswitch/bin/lame, /tmp/foo.wav, /tmp/foo.mp3, - S], [/* 16 vars */]) = 0 From FS execve(/usr/local/freeswitch/bin/lame, [/usr/local/freeswitch/bin/lame, /tmp/foo.wav, /tmp/fooo.mp3, -S], [/* 14 vars */]) = 0 I am attaching the full straces in case they are of any help. Not sure if this deserves a jira /aep -- Stopping junk mailers is good for the environment maybe it's writing some err to stderr that is being suppressed somehow On Tue, Aug 25, 2009 at 3:46 PM, Alberto Escudero-Pascual (lists) aep.li...@it46.se wrote: Hi Brian, From the CLI freeswi...@open46 system /usr/local/freeswitch/bin/lame -V2 /tmp/foo.wav /tmp/foo.mp3 -S 2009-08-25 22:41:51.556484 [NOTICE] mod_commands.c:3386 Executing command: /usr/local/freeswitch/bin/lame -V2 /tmp/foo.wav /tmp/foo.mp3 -S API CALL [system(/usr/local/freeswitch/bin/lame -V2 /tmp/foo.wav /tmp/foo.mp3 -S)] output: +OK open46:/tmp# ls foo.wav and running the command from the command line: open46:/tmp# /usr/local/freeswitch/bin/lame -V2 /tmp/foo.wav /tmp/foo.mp3 -Sopen46:/tmp# ls foo.mp3 foo.wav If I do the same with lame397 freeswi...@open46 system /usr/local/freeswitch/bin/lame397 -V2 /tmp/foo.wav /tmp/foo.mp3 -S 2009-08-25 22:44:32.743998 [NOTICE] mod_commands.c:3386 Executing command: /usr/local/freeswitch/bin/lame397 -V2 /tmp/foo.wav /tmp/foo.mp3 -S API CALL [system(/usr/local/freeswitch/bin/lame397 -V2 /tmp/foo.wav /tmp/foo.mp3 -S)] output: +OK open46:/tmp# ls foo.mp3 foo.wav Highly paranormal! Sorry for hijacking the previous thread. /aep -- Stopping junk mailers is good for the environment Try running it at the CLI and see if you see any errors. Also please do not hijack threads. The original thread [Freeswitch-users] XML- RPC on different ip than 0.0.0.0 which was hijacked by clicking reply, changing the subject and clicking send. Please in the future do not do that as it clutters up the threading and could get your query lost in the noise. Thanks, Brian On Aug 25, 2009, at 1:54 AM, Alberto Escudero-Pascual (lists) wrote: Here it comes the mystery. I am use lame 3.98.2 the mp3 file never appears, if I use version 3.97 (older version), it does!. ___ FreeSWITCH-users mailing list 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@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/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_miness...@hotmail.com MSN %3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.comPAYPAL %3aanthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org sip %3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.orggoogletalk%3Aconf %2b...@conference.freeswitch.org pstn:213-799-1400 ___ FreeSWITCH-users mailing list 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 lame_strace.txt___ FreeSWITCH-users mailing list 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
Re: [Freeswitch-users] opal build error
Thanks, will try later. On Aug 16, 2009, at 2:20 PM, Peter Olsson wrote: Make sure to do a complete rebuild. And also read the comments in jira MODOPAL-10. /Peter On 09-08-16 03.51, Seven Du dujinf...@gmail.com wrote: Hi, According to wiki it still in development status, but should compile right? Any idea about this? thanks. make In file included from mod_opal.cpp:25: mod_opal.h:151: error: conflicting return type specified for 'virtual OpalLocalConnection* FSEndPoint::CreateConnection(OpalCall, void*)' /usr/include/opal/opal/localep.h:267: error: overriding 'virtual ptlib_virtual_function_changed_or_removed** OpalLocalEndPoint::CreateConnection(OpalCall, void*)' mod_opal.cpp: In constructor 'FSConnection::FSConnection(OpalCall, FSEndPoint, switch_caller_profile_t*, switch_core_session_t*, switch_channel_t*)': mod_opal.cpp:564: error: no matching function for call to 'OpalLocalConnection::OpalLocalConnection(OpalCall, FSEndPoint, NULL)' /usr/include/opal/opal/localep.h:290: note: candidates are: OpalLocalConnection::OpalLocalConnection(OpalCall, OpalLocalEndPoint, void*, unsigned int, OpalConnection::StringOptions*, char)/usr/include/opal/opal/localep.h: 276: note: OpalLocalConnection::OpalLocalConnection(const OpalLocalConnection) mod_opal.cpp: In member function 'switch_status_t FSConnection::receive_message(switch_core_session_message_t*)': mod_opal.cpp:1037: error: 'SWITCH_CHANNEL_SESSION_LOG' was not declared in this scope make[1]: *** [mod_opal.lo] Error 1 make: *** [all] Error 1 ___ FreeSWITCH-users mailing list 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 !DSPAM:4a8767f232931167913993! ___ FreeSWITCH-users mailing list 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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Skypiax, Skype endpoint and trunk, robustness patch
Great works. I tested and reported results in jira. And as I noticed you removed the sequential line hunting methods. Though I don't use that I think someone else may need that. Think about the guy want skypeout accounts in a round robin manner, others might use that in a priority manner, that means sequential search. So it maybe a good idea to add it back. At the same time I think ANY is not a good name for that method, so named it SEQ maybe better. Then ANY can be removed later by announcing here or keep there for backwards compatibility. Thank you very much for merging in the sk list with statistic patch, and for the other two features, I found it's a little hard to split codes, so, can make two jira, and upload one patch file? Two features are: continue load on fail: make sure the module continue load even it failed to talk to a skype instance auto skype user: get the user name by the returned CURRENTUSERHANDLE other than from the config xml, for easier config. Thanks. -7- On Aug 15, 2009, at 1:43 AM, Giovanni Maruzzelli wrote: Hi FreeSWITCHers, all the users of mod_skypiax are kindly requested to test the svn trunk 14519. It contains a lot of changes meant to add stability and robustness, toward a production environment. Let me know how your feelings, and please add to the Jira any possible bug/issue/etc. Thanks to you all, -giovanni Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 ___ FreeSWITCH-users mailing list 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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Skypiax, Skype endpoint and trunk, robustness patch
On Aug 16, 2009, at 12:02 AM, Giovanni Maruzzelli wrote: On Sat, Aug 15, 2009 at 5:41 PM, Seven Dudujinf...@gmail.com wrote: And as I noticed you removed the sequential line hunting methods. Because was broken. So, I aliased it to the RR. If you think it can be useful, add a Jira for it Ok, as you think it's broken, better to leave it as is. Thank you very much for merging in the sk list with statistic patch, Thanks to you for sending the patch! I've only added the callflow of the skype client to it I think maybe you merged in an old version of the patch, the last should be skype_status_new1.diff, which I added one more statistic line and the unsigned long should be uint32_t I think. however, I can add another patch if you think it's useful. Oh, one more, it also set channel name to skypiax/RR/sk_1/other_skype_name for easy check in log. (As I also would like to see sofia/external/ to be sofia/gateway/ gw/ :) ). Two features are: continue load on fail: make sure the module continue load even it failed to talk to a skype instance h, I'm too conservative for this one: I prefer that if you configured a skype instance, you expect it to work, so the module must fail if there is not such instance One of the reason I think it's useful is one can configure to load everything on server boot time. I run two skypiax servers, one can start 20 instances in batch without any problem but the other only starts 50%, then I need manually start them over and over until I confirmed all works with client or skypiax_auth. The two servers are not in the same datacenter but all have public ip. Once it started working we never never experienced a skype instance stoped working. But I experienced that kill a skype instance immediately caused skypiax core dump. Sure it might cause other bugs even it is configured to false by default. I gona merge in my branch in case others using that. auto skype user: get the user name by the returned CURRENTUSERHANDLE other than from the config xml, for easier config. the username returned by CURRENTUSERHANDLE is checked against the config file because is the only way you can associate interface_name with its related Skype client instance on Windoz (no multiple X servers there). It by default disabled so I guess nothing will break. Thanks a lot for all your efforts!!! -giovanni On Aug 15, 2009, at 1:43 AM, Giovanni Maruzzelli wrote: Hi FreeSWITCHers, all the users of mod_skypiax are kindly requested to test the svn trunk 14519. It contains a lot of changes meant to add stability and robustness, toward a production environment. Let me know how your feelings, and please add to the Jira any possible bug/issue/etc. Thanks to you all, -giovanni Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 ___ FreeSWITCH-users mailing list 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@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@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@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] opal build error
Hi, According to wiki it still in development status, but should compile right? Any idea about this? thanks. make In file included from mod_opal.cpp:25: mod_opal.h:151: error: conflicting return type specified for ‘virtual OpalLocalConnection* FSEndPoint::CreateConnection(OpalCall, void*)’ /usr/include/opal/opal/localep.h:267: error: overriding ‘virtual ptlib_virtual_function_changed_or_removed** OpalLocalEndPoint::CreateConnection(OpalCall, void*)’ mod_opal.cpp: In constructor ‘FSConnection::FSConnection(OpalCall, FSEndPoint, switch_caller_profile_t*, switch_core_session_t*, switch_channel_t*)’: mod_opal.cpp:564: error: no matching function for call to ‘OpalLocalConnection::OpalLocalConnection(OpalCall, FSEndPoint, NULL)’ /usr/include/opal/opal/localep.h:290: note: candidates are: OpalLocalConnection::OpalLocalConnection(OpalCall, OpalLocalEndPoint, void*, unsigned int, OpalConnection::StringOptions*, char)/usr/include/opal/opal/localep.h: 276: note: OpalLocalConnection::OpalLocalConnection(const OpalLocalConnection) mod_opal.cpp: In member function ‘switch_status_t FSConnection::receive_message(switch_core_session_message_t*)’: mod_opal.cpp:1037: error: ‘SWITCH_CHANNEL_SESSION_LOG’ was not declared in this scope make[1]: *** [mod_opal.lo] Error 1 make: *** [all] Error 1 ___ FreeSWITCH-users mailing list 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] 403 forbidden on files.freeswitch.org
Got this: Forbidden You don't have permission to access /cluecon_2009/presentations/ Dale_Building_FreeSWITCH_App_Lua.pptx on this server. Apache/2.2.3 (CentOS) Server at files-sync.freeswitch.org Port 80 $ ping files.freeswitch.org PING filessync.freeswitch.netdna-cdn.com (69.174.57.101): 56 data bytes 64 bytes from 69.174.57.101: icmp_seq=0 ttl=46 time=224.900 ms 64 bytes from 69.174.57.101: icmp_seq=1 ttl=46 time=221.597 ms ^C $ ping files-sync.freeswitch.org PING www-01.freeswitch.org (216.82.231.69): 56 data bytes 64 bytes from 216.82.231.69: icmp_seq=0 ttl=44 time=271.147 ms 64 bytes from 216.82.231.69: icmp_seq=1 ttl=44 time=273.161 ms ^C ___ FreeSWITCH-users mailing list 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
Re: [Freeswitch-users] problem when adding more extension
On Aug 12, 2009, at 2:44 PM, Tzury Bar Yochay wrote:
Hi,
I wanted to add more extension to freeswitch.
to add extension 1050 with password 1234 I did the following:
$ cd /usr/local/freeswitch/conf/directory/default
created 1050.xml having all '1000' strings replaced by '1050' by
typing
$ sed s/1000/1050/g 1000.xml 1050.xml
just run reloadxml should be ok no need to rescan the profile
rescan and reload the xml by typing into the CLI
freeswi...@internal sofia profile internal rescan reloadxml
However, when I tried to login with these credentials I got the
following in the fs_cli:
2009-08-12 06:35:01 [DEBUG] sofia_reg.c:1469 sofia_reg_parse_auth()
SIP username 1050 does not match auth username
2009-08-12 06:35:01 [DEBUG] sofia_reg.c:869
sofia_reg_handle_register() Send challenge for
[1...@server_address.net]
below are the content of 1000 and 1050 xml files
please advise.
$ cat 1050.xml
include
user id=1050 mailbox=1050
params
param name=password value=1234/
param name=vm-password value=1050/
/params
variables
variable name=toll_allow value=domestic,international,local/
variable name=accountcode value=1050/
variable name=user_context value=default/
variable name=effective_caller_id_name value=Extension 1050/
variable name=effective_caller_id_number value=1050/
variable name=outbound_caller_id_name
value=$${outbound_caller_name}/
variable name=outbound_caller_id_number
value=$${outbound_caller_id}/
variable name=callgroup value=techsupport/
/variables
/user
/include
$ cat 1000.xml
include
user id=1000 mailbox=1000
params
param name=password value=1234/
param name=vm-password value=1000/
/params
variables
variable name=toll_allow value=domestic,international,local/
variable name=accountcode value=1000/
variable name=user_context value=default/
variable name=effective_caller_id_name value=Extension 1000/
variable name=effective_caller_id_number value=1000/
variable name=outbound_caller_id_name
value=$${outbound_caller_name}/
variable name=outbound_caller_id_number
value=$${outbound_caller_id}/
variable name=callgroup value=techsupport/
/variables
/user
/include
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Re: [Freeswitch-users] answer command
It's not Eyebeam but FS hung up the call because it have nothing to do after answer. You should either playback a sound, do the echo command, record, hold the call, bridge to another channel or transfer somewhere else. On Aug 12, 2009, at 4:54 PM, Maxim Tsvetov wrote: I've tried to use answer command from outbound event socket and it's working, but the problem is that FS answering the call, but SIP Client (we tried this with EyeBeam and CISCO 7960) doesn't know that call was answered. So, as long as FS doesn't know what to do with this number it then disconnects the call. 2009-08-12 11:25:07.599250 [NOTICE] mod_dptools.c:649 Channel [sofia/internal/sip:1...@10.107.181.160:42840] has been answered 2009-08-12 11:25:07.599250 [NOTICE] switch_ivr_originate.c:2015 Channel [sofia/internal/1...@10.107.249.12] has been answered 2009-08-12 11:25:07.614875 [ERR] switch_core_io.c:118 sofia/internal/sip:1...@10.107.181.160:42840 has no read codec. 2009-08-12 11:25:07.614875 [NOTICE] switch_ivr_bridge.c:503 Hangup sofia/internal/sip:1...@10.107.181.160:42840 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] 2009-08-12 11:25:07.614875 [NOTICE] switch_ivr_bridge.c:1016 Hangup sofia/internal/1...@10.107.249.12 [CS_EXECUTE] [NORMAL_CLEARING] 2009-08-12 11:25:07.630500 [NOTICE] switch_core_session.c:1086 Session 133 (sofia/internal/sip:1...@10.107.181.160:42840) Ended Maybe there is the way to acknowledge SIP client that call was answered? Regards, Maxim Tsvetov Diego Viola wrote: I suggest that you learn the differences between mod_commands commands and mod_dptools applications, and also the interfaces where you can access and use them. As said before, mod_dptools is accessible from dialplan, event socket outbound, etc. and mod_commands is accessible from the CLI, event socket (inbound/outbound), XML RPC, etc. That's all described in the wiki I think. Let us know if you have any questions =D. On Tue, Aug 11, 2009 at 2:10 PM, Diego Viola diego.vi...@gmail.com wrote: Michael, you're welcome :). Milena, answer is a mod_dptools command, you can use it from the XML dialplan or from the event socket outbound. mod_commands API are APIs that you execute from the socket, event socket inbound, etc. But you can also execute them from event socket outbound using the api command. I hope that makes sense, correct me if I'm wrong =D. On Tue, Aug 11, 2009 at 1:09 PM, Michael Collins m...@freeswitch.orgwrote: On Tue, Aug 11, 2009 at 9:05 AM, Milena testeado...@gmail.com wrote: Hello Brian, I wanna fix the wiki, but to make sure i got it right, does it only work on outbound event socket? or is there any other scenario where it would work. FYI, Diego Viola fixed the wiki. (Thanks Diego!) -MC ___ FreeSWITCH-users mailing list 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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- View this message in context: http://www.nabble.com/answer-command-tp24912812p24931876.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ FreeSWITCH-users mailing list 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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] answer command
answer only works on outbound event socket. why you don't answer in a dialplan? what's scenario you use this? On Aug 11, 2009, at 9:31 PM, Maxim Tsvetov wrote: I've tried all this command from FS console and all of them return Unknown command daqiang wang wrote: why not use: session:answer() 2009/8/11 Maxim Tsvetov maxim.tsve...@gmail.com Thank you. Is it possible to use this command from command line or via event_socket interface? Szymon Olko wrote: Maxim Tsvetov pisze: Hello, I found in WIKI list of supported commands http://wiki.freeswitch.org/wiki/Mod_commands. I need command answer but it doesn't exist (Freeswitch 1.04, Win32) That's what Freeswitch replies for this command: answer: Command not found! Сould you please help? Answer is in dialplan commands. http://wiki.freeswitch.org/wiki/Mod_dptools Szymon ___ FreeSWITCH-users mailing list 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 -- View this message in context: http://www.nabble.com/answer-command-tp24912812p24916765.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ FreeSWITCH-users mailing list 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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- View this message in context: http://www.nabble.com/answer-command-tp24912812p24917706.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ FreeSWITCH-users mailing list 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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] skypiax on Mac OS X
2009/8/10 Ivan C Myrvold i...@myrvold.org Seven, I am afraid I will not be able to help you much with the Carbon code, as I am only good at Cocoa programming. You said you chose Carbon because you only needed low level API, and that is fair enough, but I will also add that you can do the same with only linking to the Foundation framework in Cocoa. Link with Cocoa is OK but just need extra hack to the standard Makefile and not so necessary. And I think Carbon code is more C friendly. I made a jira to Skype-Dev, hope someone can help us. I looked a little at the diff file yesterday, and will investigate more today, to try to understand how you have done the Skype integration to the Freeswitch in the Carbon code. And I am glad that someone have contributed to get skypiax working in OS X. Great work so far! Ivan Den 9. aug.. 2009 kl. 20:02 skrev Seven Du: Ivan, Good to know you are a cocoa dev. Unable to check in code right now, will send the diff to you offlist for now. 0) I'm not familiar with Mac dev, just tried my best 1) It doesn't work yet, but should be able to compile, sure you already have the Skype framework in place :) 2) if run the skype delegate from a threat, then cannot get event callback. e.g. mac_client.c works but mac_client2.c doesn't. Since skypiax is running in a thread, we need to figure out this first. 3) it uses Carbon, since I think we only need to low level api, no need to bother the complicate of Cocoa. 4) strsep shows some warning on compile, haven't figured out why 5) perhaps you should only add one interface in skypiax.conf.xml 6) do you want to run multi-instances like on Linux? 7) I really not sure if it will work or not :) Let me know if it helps. I bet you can make it work. Also code will be in my branch soon. 7. On Aug 9, 2009, at 11:34 PM, Ivan C Myrvold wrote: Yes, I am interested in this, and if you have any source I could have a look at it. Ivan Den 9. aug.. 2009 kl. 17:24 skrev Seven Du: On Aug 9, 2009, at 11:10 PM, Giovanni Maruzzelli wrote: Ciao Ivan, it seems that you do not have the libX11 **development** package installed. Unfortunately I don't know about OSX, so I cannot help you, but many on the list know. BTW: it will probably be of no use to you to compile mod_skypiax on OSX, because Skype for MACOSX works in another way than Skype for Linux. That's right. If you know about MacOSX programming, please have a look at https://developer.skype.com/Docs/ApiDoc/Skype_API_on_Mac it would probably be simple enough to add a message pump for MacOSX. -giovanni Giovanni, I have a Mac and tried to get this work yesterday, but haven't got it work. Will try further if I have time. However, I don't think it's so useful because I don't know how to run and hence control multi-skype instances on Mac. If someone interested to try this I can check the code into my branch. Sincerely, Giovanni Maruzzelli = www.celliax.org via Pierlombardo 9, 20135 Milano Italy gmaruzz at celliax dot org Cell : +39-347-2665618 Fax : +39-02-87390039 On Sun, Aug 9, 2009 at 4:52 PM, Ivan C Myrvoldi...@myrvold.org wrote: I tried to compile mod_skypiax, and am getting problem with X11. On OS X Leopard, X11 is installed in /usr/X11/lib/ See below. What can I do to get past this error? I can also let you ssh into my machine. Contact me off list in case. Ivan making all mod_skypiax Compiling skypiax_protocol.c... Compiling mod_skypiax.c... mkdir .libs Compiling mod_skypiax.c ... Creating mod_skypiax.so... ld: library not found for -lX11 collect2: ld returned 1 exit status gcc -DSKYPIAX_SVN_VERSION=\14471\ -I/Users/imyrvold/Documents/ Freeswitch/freeswitch.09-08-09/src/include -I/Users/imyrvold/ Documents/ Freeswitch/freeswitch.09-08-09/libs/libteletone/src -Werror - fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 - g - ggdb -DMACOSX -g -O2 -Wall -std=c99 -pedantic -D_GNU_SOURCE - shared - o .libs/mod_skypiax.so -dynamic -bundle -force-flat- namespace .libs/ mod_skypiax.o skypiax_protocol.o /Users/imyrvold/Documents/ Freeswitch/ freeswitch.09-08-09/.libs/libfreeswitch.dylib -L/usr/lib -L/Users/ imyrvold/Documents/Freeswitch/freeswitch.09-08-09/libs/apr-util/ xml/ expat/lib /Users/imyrvold/Documents/Freeswitch/freeswitch. 09-08-09/ libs/apr-util/xml/expat/lib/.libs/libexpat.a /usr/lib/ libiconv.dylib / Users/imyrvold/Documents/Freeswitch/freeswitch.09-08-09/libs/ apr/.libs/ libapr-1.a -ldl -lpthread -lm -L/opt/local/lib -lssl -lcrypto - lz - lncurses -lX11 make[5]: *** [mod_skypiax.so] Error 1 make[4]: *** [all] Error 1 make[3]: *** [mod_skypiax-all] Error 1 make[2]: *** [all-recursive] Error 1 Den 6. aug.. 2009 kl. 18:37 skrev Giovanni Maruzzelli: No, it needs
Re: [Freeswitch-users] skypiax on Mac OS X
On Aug 9, 2009, at 11:10 PM, Giovanni Maruzzelli wrote: Ciao Ivan, it seems that you do not have the libX11 **development** package installed. Unfortunately I don't know about OSX, so I cannot help you, but many on the list know. BTW: it will probably be of no use to you to compile mod_skypiax on OSX, because Skype for MACOSX works in another way than Skype for Linux. That's right. If you know about MacOSX programming, please have a look at https://developer.skype.com/Docs/ApiDoc/Skype_API_on_Mac it would probably be simple enough to add a message pump for MacOSX. -giovanni Giovanni, I have a Mac and tried to get this work yesterday, but haven't got it work. Will try further if I have time. However, I don't think it's so useful because I don't know how to run and hence control multi-skype instances on Mac. If someone interested to try this I can check the code into my branch. Sincerely, Giovanni Maruzzelli = www.celliax.org via Pierlombardo 9, 20135 Milano Italy gmaruzz at celliax dot org Cell : +39-347-2665618 Fax : +39-02-87390039 On Sun, Aug 9, 2009 at 4:52 PM, Ivan C Myrvoldi...@myrvold.org wrote: I tried to compile mod_skypiax, and am getting problem with X11. On OS X Leopard, X11 is installed in /usr/X11/lib/ See below. What can I do to get past this error? I can also let you ssh into my machine. Contact me off list in case. Ivan making all mod_skypiax Compiling skypiax_protocol.c... Compiling mod_skypiax.c... mkdir .libs Compiling mod_skypiax.c ... Creating mod_skypiax.so... ld: library not found for -lX11 collect2: ld returned 1 exit status gcc -DSKYPIAX_SVN_VERSION=\14471\ -I/Users/imyrvold/Documents/ Freeswitch/freeswitch.09-08-09/src/include -I/Users/imyrvold/ Documents/ Freeswitch/freeswitch.09-08-09/libs/libteletone/src -Werror - fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g - ggdb -DMACOSX -g -O2 -Wall -std=c99 -pedantic -D_GNU_SOURCE -shared - o .libs/mod_skypiax.so -dynamic -bundle -force-flat-namespace .libs/ mod_skypiax.o skypiax_protocol.o /Users/imyrvold/Documents/ Freeswitch/ freeswitch.09-08-09/.libs/libfreeswitch.dylib -L/usr/lib -L/Users/ imyrvold/Documents/Freeswitch/freeswitch.09-08-09/libs/apr-util/xml/ expat/lib /Users/imyrvold/Documents/Freeswitch/freeswitch.09-08-09/ libs/apr-util/xml/expat/lib/.libs/libexpat.a /usr/lib/ libiconv.dylib / Users/imyrvold/Documents/Freeswitch/freeswitch.09-08-09/libs/ apr/.libs/ libapr-1.a -ldl -lpthread -lm -L/opt/local/lib -lssl -lcrypto -lz - lncurses -lX11 make[5]: *** [mod_skypiax.so] Error 1 make[4]: *** [all] Error 1 make[3]: *** [mod_skypiax-all] Error 1 make[2]: *** [all-recursive] Error 1 Den 6. aug.. 2009 kl. 18:37 skrev Giovanni Maruzzelli: No, it needs implementation of the message pump between the module and the Skype API. It's probably kind of trivial, if no other problems I'm not aware of. I do not have a Mac to implement it, tough :-(. -giovanni Sincerely, Giovanni Maruzzelli = www.celliax.org via Pierlombardo 9, 20135 Milano Italy gmaruzz at celliax dot org Cell : +39-347-2665618 Fax : +39-02-87390039 On Thu, Aug 6, 2009 at 5:55 PM, Brian Westbr...@freeswitch.org wrote: I'm not sure about that one I haven't tried lately because the API differs on the Mac last I looked at it. /b On Aug 6, 2009, at 10:53 AM, Ivan C Myrvold wrote: Is skypiax now working on Mac OS X in Freeswitch? Ivan ___ FreeSWITCH-users mailing list 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@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@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@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] skypiax on Mac OS X
Ivan, Good to know you are a cocoa dev. Unable to check in code right now, will send the diff to you offlist for now. 0) I'm not familiar with Mac dev, just tried my best 1) It doesn't work yet, but should be able to compile, sure you already have the Skype framework in place :) 2) if run the skype delegate from a threat, then cannot get event callback. e.g. mac_client.c works but mac_client2.c doesn't. Since skypiax is running in a thread, we need to figure out this first. 3) it uses Carbon, since I think we only need to low level api, no need to bother the complicate of Cocoa. 4) strsep shows some warning on compile, haven't figured out why 5) perhaps you should only add one interface in skypiax.conf.xml 6) do you want to run multi-instances like on Linux? 7) I really not sure if it will work or not :) Let me know if it helps. I bet you can make it work. Also code will be in my branch soon. 7. On Aug 9, 2009, at 11:34 PM, Ivan C Myrvold wrote: Yes, I am interested in this, and if you have any source I could have a look at it. Ivan Den 9. aug.. 2009 kl. 17:24 skrev Seven Du: On Aug 9, 2009, at 11:10 PM, Giovanni Maruzzelli wrote: Ciao Ivan, it seems that you do not have the libX11 **development** package installed. Unfortunately I don't know about OSX, so I cannot help you, but many on the list know. BTW: it will probably be of no use to you to compile mod_skypiax on OSX, because Skype for MACOSX works in another way than Skype for Linux. That's right. If you know about MacOSX programming, please have a look at https://developer.skype.com/Docs/ApiDoc/Skype_API_on_Mac it would probably be simple enough to add a message pump for MacOSX. -giovanni Giovanni, I have a Mac and tried to get this work yesterday, but haven't got it work. Will try further if I have time. However, I don't think it's so useful because I don't know how to run and hence control multi-skype instances on Mac. If someone interested to try this I can check the code into my branch. Sincerely, Giovanni Maruzzelli = www.celliax.org via Pierlombardo 9, 20135 Milano Italy gmaruzz at celliax dot org Cell : +39-347-2665618 Fax : +39-02-87390039 On Sun, Aug 9, 2009 at 4:52 PM, Ivan C Myrvoldi...@myrvold.org wrote: I tried to compile mod_skypiax, and am getting problem with X11. On OS X Leopard, X11 is installed in /usr/X11/lib/ See below. What can I do to get past this error? I can also let you ssh into my machine. Contact me off list in case. Ivan making all mod_skypiax Compiling skypiax_protocol.c... Compiling mod_skypiax.c... mkdir .libs Compiling mod_skypiax.c ... Creating mod_skypiax.so... ld: library not found for -lX11 collect2: ld returned 1 exit status gcc -DSKYPIAX_SVN_VERSION=\14471\ -I/Users/imyrvold/Documents/ Freeswitch/freeswitch.09-08-09/src/include -I/Users/imyrvold/ Documents/ Freeswitch/freeswitch.09-08-09/libs/libteletone/src -Werror - fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 - g - ggdb -DMACOSX -g -O2 -Wall -std=c99 -pedantic -D_GNU_SOURCE - shared - o .libs/mod_skypiax.so -dynamic -bundle -force-flat- namespace .libs/ mod_skypiax.o skypiax_protocol.o /Users/imyrvold/Documents/ Freeswitch/ freeswitch.09-08-09/.libs/libfreeswitch.dylib -L/usr/lib -L/Users/ imyrvold/Documents/Freeswitch/freeswitch.09-08-09/libs/apr-util/ xml/ expat/lib /Users/imyrvold/Documents/Freeswitch/freeswitch.09-08-09/ libs/apr-util/xml/expat/lib/.libs/libexpat.a /usr/lib/ libiconv.dylib / Users/imyrvold/Documents/Freeswitch/freeswitch.09-08-09/libs/ apr/.libs/ libapr-1.a -ldl -lpthread -lm -L/opt/local/lib -lssl -lcrypto -lz - lncurses -lX11 make[5]: *** [mod_skypiax.so] Error 1 make[4]: *** [all] Error 1 make[3]: *** [mod_skypiax-all] Error 1 make[2]: *** [all-recursive] Error 1 Den 6. aug.. 2009 kl. 18:37 skrev Giovanni Maruzzelli: No, it needs implementation of the message pump between the module and the Skype API. It's probably kind of trivial, if no other problems I'm not aware of. I do not have a Mac to implement it, tough :-(. -giovanni Sincerely, Giovanni Maruzzelli = www.celliax.org via Pierlombardo 9, 20135 Milano Italy gmaruzz at celliax dot org Cell : +39-347-2665618 Fax : +39-02-87390039 On Thu, Aug 6, 2009 at 5:55 PM, Brian Westbr...@freeswitch.org wrote: I'm not sure about that one I haven't tried lately because the API differs on the Mac last I looked at it. /b On Aug 6, 2009, at 10:53 AM, Ivan C Myrvold wrote: Is skypiax now working on Mac OS X in Freeswitch? Ivan ___ FreeSWITCH-users mailing list 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
Re: [Freeswitch-users] skypiax on Mac OS X
On Aug 9, 2009, at 11:52 PM, Giovanni Maruzzelli wrote: Seven, thanks a lot for your effort, please let your stuff be available, maybe Ivan can make use of it! svn diff http://svn.freeswitch.org/svn/freeswitch/branches/seven -r 14473 :14475 When this done I think it's better to split codes into skypiax_protocol.c skypiax_protocol_mac.c skypiax_protocol_linux.c skypiax_protocol_windows.c :) 7. Ivan, in the file src/mod/endpoints/mod_skypiax/skypiax_protocol.c add you will find #ifdef WIN32 . it conditional compiles code between WIN32 and linux. You need to add another #ifdef, so it will compile for OSX. You will probably be able to use the same pipe mechanism as in Linux (normal POSIX pipes). You will for sure need to implement the part that deals with the Skype API. Maybe it will be not much more than reusing the example code to interact with the API. Please, let us know how it goes, and feel *very* free to ask for further info. -giovanni Sincerely, Giovanni Maruzzelli = www.celliax.org via Pierlombardo 9, 20135 Milano Italy gmaruzz at celliax dot org Cell : +39-347-2665618 Fax : +39-02-87390039 On Sun, Aug 9, 2009 at 5:34 PM, Ivan C Myrvoldi...@myrvold.org wrote: Yes, I am interested in this, and if you have any source I could have a look at it. Ivan Den 9. aug.. 2009 kl. 17:24 skrev Seven Du: On Aug 9, 2009, at 11:10 PM, Giovanni Maruzzelli wrote: Ciao Ivan, it seems that you do not have the libX11 **development** package installed. Unfortunately I don't know about OSX, so I cannot help you, but many on the list know. BTW: it will probably be of no use to you to compile mod_skypiax on OSX, because Skype for MACOSX works in another way than Skype for Linux. That's right. If you know about MacOSX programming, please have a look at https://developer.skype.com/Docs/ApiDoc/Skype_API_on_Mac it would probably be simple enough to add a message pump for MacOSX. -giovanni Giovanni, I have a Mac and tried to get this work yesterday, but haven't got it work. Will try further if I have time. However, I don't think it's so useful because I don't know how to run and hence control multi-skype instances on Mac. If someone interested to try this I can check the code into my branch. Sincerely, Giovanni Maruzzelli = www.celliax.org via Pierlombardo 9, 20135 Milano Italy gmaruzz at celliax dot org Cell : +39-347-2665618 Fax : +39-02-87390039 On Sun, Aug 9, 2009 at 4:52 PM, Ivan C Myrvoldi...@myrvold.org wrote: I tried to compile mod_skypiax, and am getting problem with X11. On OS X Leopard, X11 is installed in /usr/X11/lib/ See below. What can I do to get past this error? I can also let you ssh into my machine. Contact me off list in case. Ivan making all mod_skypiax Compiling skypiax_protocol.c... Compiling mod_skypiax.c... mkdir .libs Compiling mod_skypiax.c ... Creating mod_skypiax.so... ld: library not found for -lX11 collect2: ld returned 1 exit status gcc -DSKYPIAX_SVN_VERSION=\14471\ -I/Users/imyrvold/Documents/ Freeswitch/freeswitch.09-08-09/src/include -I/Users/imyrvold/ Documents/ Freeswitch/freeswitch.09-08-09/libs/libteletone/src -Werror - fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 - g - ggdb -DMACOSX -g -O2 -Wall -std=c99 -pedantic -D_GNU_SOURCE - shared - o .libs/mod_skypiax.so -dynamic -bundle -force-flat- namespace .libs/ mod_skypiax.o skypiax_protocol.o /Users/imyrvold/Documents/ Freeswitch/ freeswitch.09-08-09/.libs/libfreeswitch.dylib -L/usr/lib -L/Users/ imyrvold/Documents/Freeswitch/freeswitch.09-08-09/libs/apr-util/ xml/ expat/lib /Users/imyrvold/Documents/Freeswitch/freeswitch. 09-08-09/ libs/apr-util/xml/expat/lib/.libs/libexpat.a /usr/lib/ libiconv.dylib / Users/imyrvold/Documents/Freeswitch/freeswitch.09-08-09/libs/ apr/.libs/ libapr-1.a -ldl -lpthread -lm -L/opt/local/lib -lssl -lcrypto - lz - lncurses -lX11 make[5]: *** [mod_skypiax.so] Error 1 make[4]: *** [all] Error 1 make[3]: *** [mod_skypiax-all] Error 1 make[2]: *** [all-recursive] Error 1 Den 6. aug.. 2009 kl. 18:37 skrev Giovanni Maruzzelli: No, it needs implementation of the message pump between the module and the Skype API. It's probably kind of trivial, if no other problems I'm not aware of. I do not have a Mac to implement it, tough :-(. -giovanni Sincerely, Giovanni Maruzzelli = www.celliax.org via Pierlombardo 9, 20135 Milano Italy gmaruzz at celliax dot org Cell : +39-347-2665618 Fax : +39-02-87390039 On Thu, Aug 6, 2009 at 5:55 PM, Brian Westbr...@freeswitch.org wrote: I'm not sure about that one I haven't tried lately because the API differs on the Mac last I looked at it. /b On Aug 6, 2009, at 10:53 AM, Ivan C Myrvold wrote: Is skypiax now working on Mac OS X
Re: [Freeswitch-users] Softphone control
You can run FreeSWITCH as a softphone and control it. http://wiki.freeswitch.org/wiki/Freeswitch_softphone 2009/8/7 Artem Vasiliev ryde...@googlemail.com Hi I have FreeSwitch and external application, which communicates to it via event socket - listens for events for certain number and gives some commands. Is it possible for this application to control client softphones, for example, make them answer or hold, using the event socket or other FreeSwitch capabilities? ___ FreeSWITCH-users mailing list 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@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] A few questions about lua
ALL- I have a few questions when scripting lua. According to wiki, it is possible to run looping forever lua scripts through start-up config or luarun. 1) Will the lua script stop when unload mod_lua? I experienced core dump when unload mod_lua while there was a running lua script. Reported on jira. 2) How to stop a forever running lua script? I stop it by listening a CUSTOM event fired elsewhere. See code below. Is there any standard way like luastop ? 3) Any way to show how many running lua scripts? luashow ? 4) It seems cannot get the lua script name in a lua script, I made a patch to jira by assign it to the argv[0]. 5) Seems that only EventConsumer(all) working. EventConsumer(CHANNEL_HANUP CUSTOM lua::stop) doesn't seem to work. Any idea to this? Thanks a lot. code example: con = freeswitch.EventConsumer(all); argv[0] = test.lua freeswitch.consoleLog(info, Lua Script [ .. argv[0] .. ] Starting =\n); local all_events = 0 for e in (function() return con:pop(1) end) do -- freeswitch.consoleLog(info, event\n .. e:serialize(xml)); all_events = all_events + 1; freeswitch.consoleLog(info, all_events: .. all_events .. \n) event_name = e:getHeader(Event-Name) or event_subclass = e:getHeader(Event-Subclass) or if (event_name == CUSTOM and event_subclass == lua::stop) then freeswitch.consoleLog(info, -lua Script [ .. argv[0] .. ]---Exiting--\n) break end end ___ FreeSWITCH-users mailing list 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
Re: [Freeswitch-users] A few questions about lua
for e in (function() return con:pop(1) end) do btw, the script works. Thanks. On Aug 6, 2009, at 6:47 PM, Eli Hayun wrote: Hi I dont know about events so much but I cannot see variable e is setting event_name = e:getHeader(Event-Name) or event_subclass = e:getHeader(Event-Subclass) or regurds Eli On Thu, 2009-08-06 at 12:58 +0300, Seven Du wrote: ALL- I have a few questions when scripting lua. According to wiki, it is possible to run looping forever lua scripts through start-up config or luarun. 1) Will the lua script stop when unload mod_lua? I experienced core dump when unload mod_lua while there was a running lua script. Reported on jira. 2) How to stop a forever running lua script? I stop it by listening a CUSTOM event fired elsewhere. See code below. Is there any standard way like luastop ? 3) Any way to show how many running lua scripts? luashow ? 4) It seems cannot get the lua script name in a lua script, I made a patch to jira by assign it to the argv[0]. 5) Seems that only EventConsumer(all) working. EventConsumer(CHANNEL_HANUP CUSTOM lua::stop) doesn't seem to work. Any idea to this? Thanks a lot. code example: con = freeswitch.EventConsumer(all); argv[0] = test.lua freeswitch.consoleLog(info, Lua Script [ .. argv[0] .. ] Starting =\n); local all_events = 0 for e in (function() return con:pop(1) end) do -- freeswitch.consoleLog(info, event\n .. e:serialize(xml)); all_events = all_events + 1; freeswitch.consoleLog(info, all_events: .. all_events .. \n) event_name = e:getHeader(Event-Name) or event_subclass = e:getHeader(Event-Subclass) or if (event_name == CUSTOM and event_subclass == lua::stop) then freeswitch.consoleLog(info, -lua Script [ .. argv[0] .. ]---Exiting--\n) break end end ___ FreeSWITCH-users mailing list 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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] A few questions about lua
Sorry it's a typo. I read the code, it works not like in event socket. So, only works with one event. either EventConsumer(all) or EventConsumer(CUSTOM, lua::stop); Thank you. 2009/8/7 Raffaele P. Guidi raffaele.p.gu...@gmail.com 5) Seems that only EventConsumer(all) working. EventConsumer(CHANNEL_HANUP CUSTOM lua::stop) doesn't seem to work. Any idea to this? isn't it CHANNEL_HAN*G*UP? Is the G missing only in the email or in the code, too? On Thu, Aug 6, 2009 at 17:52, Seven Du dujinf...@gmail.com wrote: for e in (function() return con:pop(1) end) do btw, the script works. Thanks. On Aug 6, 2009, at 6:47 PM, Eli Hayun wrote: Hi I dont know about events so much but I cannot see variable e is setting event_name = e:getHeader(Event-Name) or event_subclass = e:getHeader(Event-Subclass) or regurds Eli On Thu, 2009-08-06 at 12:58 +0300, Seven Du wrote: ALL- I have a few questions when scripting lua. According to wiki, it is possible to run looping forever lua scripts through start-up config or luarun. 1) Will the lua script stop when unload mod_lua? I experienced core dump when unload mod_lua while there was a running lua script. Reported on jira. 2) How to stop a forever running lua script? I stop it by listening a CUSTOM event fired elsewhere. See code below. Is there any standard way like luastop ? 3) Any way to show how many running lua scripts? luashow ? 4) It seems cannot get the lua script name in a lua script, I made a patch to jira by assign it to the argv[0]. 5) Seems that only EventConsumer(all) working. EventConsumer(CHANNEL_HANUP CUSTOM lua::stop) doesn't seem to work. Any idea to this? Thanks a lot. code example: con = freeswitch.EventConsumer(all); argv[0] = test.lua freeswitch.consoleLog(info, Lua Script [ .. argv[0] .. ] Starting =\n); local all_events = 0 for e in (function() return con:pop(1) end) do -- freeswitch.consoleLog(info, event\n .. e:serialize(xml)); all_events = all_events + 1; freeswitch.consoleLog(info, all_events: .. all_events .. \n) event_name = e:getHeader(Event-Name) or event_subclass = e:getHeader(Event-Subclass) or if (event_name == CUSTOM and event_subclass == lua::stop) then freeswitch.consoleLog(info, -lua Script [ .. argv[0] .. ]---Exiting--\n) break end end ___ FreeSWITCH-users mailing list 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@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@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] FreeSWITCH code to emulate Asterisk dialing announcement
I think you can check the loopback endpoint or inline dialplan. On Aug 4, 2009, at 3:23 AM, Michael Frager wrote: Hello, I'm in the process of moving my VOIP application from Asterisk to FreeSWITCH. I was wondering if it is possible to emulate the call announcement feature that is available on Asterisk. On Asterisk it looks like this, with the A(...) parameter: Dial(SIP/1551212|180|A(connecttone1)) Note that this announcement is only played for the called party, the calling party does NOT hear the tone. I'm guessing this can be done with FreeSWITCH. Does anyone know how I might accomplish this? Thanks in advance, -Mike Fragre ___ FreeSWITCH-users mailing list 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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] How to distinguish between the different type of call rejections from Javascript?
variable_originate_disposition On Aug 5, 2009, at 9:56 PM, Max Bridgewater wrote: Hi, Say i originate a call to a mobile phone and the call fails. There are many possible reasons: congestion, user busy, call rejected by user, etc. Is there a way i can get the failure code from Javascript? Thanks, Max. ___ FreeSWITCH-users mailing list 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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Multiple DIDs per SIP trunk (how to configure?)
mod_easyroute? 2009/8/6 Vladimir Rodionov vladrodio...@gmail.com Hi, everybody This is a newbie question: Suppose I have XX (variable dynamic number) DIDs assigned to one sip trunk (from VOIP provider ABC ). All calls coming from VOIP provider ABC MUST be routed to the same lua/js/whatever script. Is it possible in FS? If yes, how everything should be configuered? Dialplan, sip gateway? One more question: suppose it is doeable as I hope then how can I get in my script CalleeID (not a CallerID)? Basicaly, I want to acomplish the following: 1. Avoid re-configuring FS every time I got new bunch of DIDs assigned/released from/to my Voip provider. 2. Have a way of extracting CalleeID in my script. TIA, Vladimir Rodionov ___ FreeSWITCH-users mailing list 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@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] In the spirit of ClueCon: Our FreeSWITCH Story
Hello All - In the spirit of ClueCon (which we are missing this year, but hopefully not next), we wanted to document our FreeSWITCH Story. We've posted it to the wiki( http://wiki.freeswitch.org/wiki/FreeSWITCH_Testimonial_on_Idapted.com) and it is copied below. Thank you all and enjoy a good conference! Seven Du (seven) Jonathan Palley (jpalley_idapted) Idapted Ltd. *How FreeSWITCH has created hundreds of job opportunities and changed lives. * We want to share our experience working with FreeSWITCH. FreeSWITCH has been a key enabler of our business. We hope this story can be a small way to say a very big THANK YOU ALL. Changing lives is an over-used cliche, but in this case, FreeSWITCH has really allowed us to do just that. What We Do: We are not a telephony business; we are an educational technology and service business. In Asia (China, in our case) students must pass English examinations to study or work abroad and gain new experiences. However, there is limited access to native English speakers and the access students can gain is typically very expensive. At the same time, in the U.S., there are many professionals looking for work-at-home opportunities - people who need jobs and would create great teachers. Through our technology and content we empower these people to be effective English teachers. Does it work? Yes. The majority of our students are getting test scores that many failed for years to get. Just hours ago one student called one of our sales agents crying with joy. And for our teachers, they are now working in an industry that was previously unavailable to those living in the U.S. http://www.idapted.com Why FreeSWITCH Enables This: FreeSWITCH has been a key enabler of our business. Recording calls, controlling routing, integrating with various web-based interfaces, enabling multiple endpoints - these are all key features of what we must do. Most importantly, setting up various servers and routes to mitigate cross-Pacific and country-specific network challenges is key. Doing what we are doing with commercial solutions would have made the business unworkable. Our Experiences with FreeSWITCH: We started using FreeSWITCH as our VoIP Platform in April 2008, after receiving unsatisfactory results with other open source solutions. It took one day of reading through the FreeSWITCH source code to know, this is it. This is the VoIP platform we build our business on. It took a few days of working with the extremely competent and focused community to re-affirm this commitment. Our Setup: Our teachers use a custom software that integrates a VoIP client with our web based platform. Students connect to our teachers on-demand. Simply put, on a web-based comet interface the student enters a phone number (or a skype name or a gtalk account) and our platform bridges the best available trainer and the student. At the same time a web-based interface is being updated. The challenge for us is the connection between teachers and students over a cross-continent network. For example, we experienced problems earlier this year when a Asis-Pacific communication fiber broken... So, we've learned to setup multi servers in multiple datacenters for redundancy. We run multi instances of FreeSWITCH so we can always use the cutting edge and mitigate the effects of bugs. A main, stable FreeSWITCH(FS) instance connect to other FreeSWITCHes - Fs-skype only loads mod_skypiax and FS-gtalk only loads mod_dingaling. Here is one beauty of FS: We just had to create different conf dirs (/usr/local/freeswitch, /usr/local/skype, /usr/local/gtalk etc). This allows us to run the same code base over different configurations, and call skype and gtalk accounts just like a normal PSTN gateway (sofia/gateway/pstn/ or sofia/gateway/skype/ or sofia/gateway/gtalk/ ). More important, if one FS (say FS-skype) behaves abnormally or crashes, we can easily change to another FS-skype server (we run other servers located in various places in China and HK for redundancy). FS --| |---PSTN gateways |--- FS-skype |--- FS-gtalk |--- FS-skype2 |--- more ... COMMUNITY: The community's commitment cannot be undervalued. The insightful, modular design of FreeSWITCH allows anyone to contribute, whereever their skills lie. It also allows us to easily make modifications to the underlying code to suit our specific use-cases We want to highlight a few key people and modules in the FS ecosystem: mod_sofia: SIP is how we connect to our PSTN gateways and to our teachers clients. PSTN is zero-conf for the user and mitigates troubles with the end users network/microphone, etc (which is significant with our user base). However, cheap providers fail randomly and FreeSWITCH's ability to control routing, use multiple endpoints all while clearly seeing what is going on is key. Most importantly, anthm and the core team have been super helpful in getting SIP to work with us. Back in the pre 1.0 days
Re: [Freeswitch-users] In the spirit of ClueCon: Our FreeSWITCH Story
And I added this on the wiki page: mod_conference and mod_fifo: We also use FreeSWITCH in our office environment as a PBX for call center and customer service connected with VoIP and PSTN(openzap) gateways. It is integrated into our CRM system naturally and just made sales process, business logic and world wide conference much more simpler and easier. :) 7 2009/8/5 Jonathan Palley jpal...@idapted.com Diego - Already done. See the bottom of the page (we linked to another page because of its length)! :) Jonathan Palley Idapted Ltd. ___ FreeSWITCH-users mailing list 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
Re: [Freeswitch-users] telnet to event socket
On Aug 3, 2009, at 10:30 PM, Ngo-Vi Hoai-Anh wrote: Hi, I'm taking a close look at event socket on FS 1.0.3. Configuration is the same as on http://wiki.freeswitch.org/wiki/Event_Socket. fs.pl and fsconsole.pl work but I was not able to telnet to port 8021. As I've done that I received somewhat like: #auth/request I typed in: auth ClueCon followed by two Enters (\n\n). After some seconds I've got the message 'connection close by foreign host' Any ideas? Thank you Hoai-Anh ___ FreeSWITCH-users mailing list 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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Missing mod_curl
On Aug 4, 2009, at 12:01 AM, afshin afzali wrote: I've gotten 1.0.4pre9 , but i can not see it :( -- afshin In trunk. On Mon, Aug 3, 2009 at 6:33 PM, Michael Jerris m...@jerris.com wrote: It is still there. On Aug 3, 2009, at 8:38 AM, afshin afzali a.afzali2...@gmail.com wrote: Hi, I'll appreciate if somebody tell me where has gone the mod_curl ? I just need to use it for http method calls. Regards, -- afshin ___ FreeSWITCH-users mailing list 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@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@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] how to enable ESL for ruby?
Hi Brian, Sorry responding late. I still cannot get this work, can you take a look? http://pastebin.freeswitch.org/9877 Everything works fine on Linux but not on my MAC. I have the default ruby framework and port install on /opt/local/bin/ruby, however, even I changed the Makefile to use the default ruby framework, it just doesn't work though the ESL.so compiled successfully. Where I'm wrong? Thanks. 2009/5/27 Brian West br...@freeswitch.org On May 26, 2009, at 11:20 AM, dujinfang wrote: Thanks Brain. Got ESL.so, however on my Mac it is #include ruby.h instead of ruby/ruby.h. Actually since we do -framework Ruby it should be ruby/ruby but I think the line above the -framework Ruby should be removed since you're doing i tthe Mac way. /b ___ FreeSWITCH-users mailing list 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
Re: [Freeswitch-users] Skypiax, Skype endpoint and trunk, revamped
Thanks for the great work. Just want you know that 20 channels with the same username works well on my server. And echo() works without any problem. An updated version of Round Robin hunt and a minor bug posted on jira. Thanks again. 2009/7/27 Giovanni Maruzzelli gmar...@celliax.org Ciao FreeSWITCHers, please have a look at the much changed wiki page: http://wiki.freeswitch.org/wiki/Skypiax_Skype_Endpoint_and_Trunk and checkout and test the code in svn. Much has happened, various bug fixes and features added. Most notable: - multiple instances of the same Skype username on Linux (eg: running 20 concurrent channels as Bob Skype user) - adding and removing interfaces on the fly (patch sent by Muhammad Shahzad) - easier creation of Skype clients configuration directory - reduced latency - better robustness - running as Windows Service - customized ALSA driver for more devices with less IRQs and context switches - custom kernel tickless and 100HZ (eg. solves high load problems in CentOS and in virtualization) - interactive command line client for prototyping Also, please note that ALSA drivers version 1.0.20 seems to be much more stable in our kind of usage (snd-dummy). Various other enhancements will come, but in the mean time please give feedback on the current svn code (we want to be robust for the 1.0.4 Release :-) ) See you all at www.cluecon.com, talk on Skypiax August 4th at 4.30 pm ! -giovanni Sincerely, Giovanni Maruzzelli = www.celliax.org via Pierlombardo 9, 20135 Milano Italy gmaruzz at celliax dot org Cell : +39-347-2665618 Fax : +39-02-87390039 ___ FreeSWITCH-users mailing list 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@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] FreeSWITCH-Air, Another GUI?
ALL, I know you guys more prefer a CLI version of softphone to a GUI version. But I still would like to share this: http://wiki.freeswitch.org/wiki/FsAir And feel free to give me feedbacks. I'v only played a few days of ActionScript, it's highly appreciated if someone can give me help on the following problems. 1) Bounce the icon on Mac on incoming call. 2) Show a small window on incoming call. 3) Is it possible to block read/write a socket? I want to implement sendRecv() in ActionScript like in the C version of ESL. Thanks. ___ FreeSWITCH-users mailing list 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
Re: [Freeswitch-users] ERROR in Sofia internal profile
chances are the tcp/udp port (5060?) already used by other software, are you running softphone on the same computer? On Jul 11, 2009, at 4:29 PM, velusamy velu wrote: Dear Friends, When I reload the mod_sofia I have got the following error. 2009-07-11 13:19:32 [ERR] sofia.c:739 sofia_profile_thread_run() Error Creating SIP UA for profile: internal Please any one explain about this error and please give any suggestions to solve this problem.. Thanks in Advance.. Regards, K.Velusamy ___ Freeswitch-users mailing list 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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Jingle (Mod_Dingaling Dialplan Extention
for outbound: extension name=test_gtalk condition field=destination_number expression=^(@gmail.com) action application=bridge data=dingaling/gmail.com/$1/ /condition /extension for inbount: you should be able to set context and extension in your dingaling profile, and set the dialplan accordingly. as always you can press F8 on console to see dialplan information On Jul 2, 2009, at 10:19 PM, Meftah Tayeb wrote: hello, i asked about mod_dingaling usage befor finally, i configured my XMPP acount with my FS, connected to it and is ready / Online now i need to call / recev call from / to GTalk (Jingle) cool anyone give me a sample Dialplan Extention? thanks __ Information from ESET NOD32 Antivirus, version of virus signature database 4209 (20090702) __ The message was checked by ESET NOD32 Antivirus. http://www.eset.com ___ Freeswitch-users mailing list 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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] how to build on MAC using /opt/local/include/tiffio.h
Thanks. I installed libtiff from source to /usr/local/ and now it works. On Jun 30, 2009, at 11:49 PM, Michael Jerris wrote: I think that detection is not working on mac right due to it looking in default search paths. I am in process of fixing this to use in tree libtiff soon so this should fix this issue. Mike On Jun 29, 2009, at 11:08 PM, seven wrote: Hi, I'm on the latest svn 14041, and I have tiff installed with port install tiff, how can I tell FS to find libtiff at /opt/local/ include? on Linux the package should be libtiff-dev of libtiff-devel, but I think tiff is the equivalent on Mac. previous versions build ok on my Mac, possible to include the libtiff in trunk like other libs? checking tiffio.h usability... no checking tiffio.h presence... no checking for tiffio.h... no checking pthread.h usability... yes checking pthread.h presence... yes checking for pthread.h... yes checking X11/X.h usability... yes checking X11/X.h presence... yes checking for X11/X.h... yes checking for libxml/xmlmemory.h... checking libxml/xmlmemory.h usability... yes checking libxml/xmlmemory.h presence... yes checking for libxml/xmlmemory.h... yes checking libxml/parser.h usability... yes checking libxml/parser.h presence... yes checking for libxml/parser.h... yes checking libxml/xinclude.h usability... yes checking libxml/xinclude.h presence... yes checking for libxml/xinclude.h... yes checking FL/Fl.H usability... no checking FL/Fl.H presence... no checking for FL/Fl.H... no checking FL/Fl_Overlay_Window.H usability... no checking FL/Fl_Overlay_Window.H presence... no checking for FL/Fl_Overlay_Window.H... no checking FL/Fl_Light_Button.H usability... no checking FL/Fl_Light_Button.H presence... no checking for FL/Fl_Light_Button.H... no checking FL/fl_draw.H usability... no checking FL/fl_draw.H presence... no checking for FL/fl_draw.H... no checking FL/Fl_Cartesian.H usability... no checking FL/Fl_Cartesian.H presence... no checking for FL/Fl_Cartesian.H... no checking FL/Fl_Audio_Meter.H usability... no checking FL/Fl_Audio_Meter.H presence... no checking for FL/Fl_Audio_Meter.H... no checking for cos in -lm... yes checking for library containing sinf... none required checking for library containing cosf... none required checking for library containing tanf... none required checking for library containing asinf... none required checking for library containing acosf... none required checking for library containing atanf... none required checking for library containing atan2f... none required checking for library containing ceilf... none required checking for library containing floorf... none required checking for library containing powf... none required checking for library containing expf... none required checking for library containing logf... none required checking for library containing log10f... none required checking for TIFFOpen in -ltiff... no configure: error: Can't build without libtiff (does your system require a libtiff-devel package?) configure: error: ./configure.gnu failed for libs/spandsp ___ Freeswitch-users mailing list 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@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] mod_dingaling no audio
3) param name=disable-rtp-auto-adjust value=true/ is not really required at least for my working setup behind the NAT router. ok, this param is originally added for another problem http://jira.freeswitch.org/browse/MODENDP-198 . But I think it might be useful for this. ___ Freeswitch-users mailing list 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
Re: [Freeswitch-users] How to change outbound_caller_name=FreeSWITCH and number=000000000
depending how do you make out going call. On Jun 23, 2009, at 2:39 PM, Edmar Cruz wrote: Actually the extension_caller_id=Extension 1001 and extension_caller_number=1001 is set as Harmeet says but the same issue FreeSwitch the caller name and the number is 000 i just want 1001 the caller number and the id Edmar Edward Q. wrote: Sorry Edmar I missundertood you .. I thought you wanted to change the number showing once you were going out not the 1001.xml file. In this case Harmeet is right. There you have those values to to make the changes. My bad. Ed On Tue, Jun 23, 2009 at 12:46 AM, Harmeet Singh harm...@litatel.com wrote: In my case the 1001 resides in - /usr/local/freeswitch/conf/directory/default/1001.xml And you set the Caller Name and ID by adding - On Tue, Jun 23, 2009 at 12:26 AM, Edward Q. q.edw...@gmail.com wrote: Edmar Eso esta en freeswitch/conf/vars.xml en ese archivo. If i am not mistaken and anyone welcome to correct me i just told Edmar this is set in freeswitch/conf/vars.xml ... file Ed On Tue, Jun 23, 2009 at 12:02 AM, Edmar Cruz darklio...@yahoo.comwrote: When I calling an outbound extension it appears: name is FreeSWITCH and number is 0 How can i change it depends on the user who is calling? Sample 1001-64521223 I just want the name 1001 to appear not FreeSWITCH same as the number Thanks -- View this message in context: http://www.nabble.com/How-to-change-outbound_caller_name%3DFreeSWITCH-and-number%3D0-tp24158823p24158823.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ Freeswitch-users mailing list 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@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@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- View this message in context: http://www.nabble.com/How-to-change-outbound_caller_name%3DFreeSWITCH-and-number%3D0-tp24158823p24160712.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ Freeswitch-users mailing list 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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] How can I join two freeswitch on two servers?
comment lines in the user directory do the trick:
variable name=effective_caller_id_name value=Extension
1000/
variable name=effective_caller_id_number value=1000/
variable name=outbound_caller_id_name value=$$
{outbound_caller_name}/
variable name=outbound_caller_id_number value=$$
{outbound_caller_id}/
On Jun 17, 2009, at 12:26 PM, Edmar Cruz wrote:
If FS A has an account 8011105 does FS B also nid to register
8011105? Yes it
working on a gateway but the username of the gateway was shown on my
softphone and also it nids a password for the gateway... is there an
option
to view the caller name and number of the FS A gateway to FS B?
Brian West-3 wrote:
COPY paste fail :)
something like that as per the example.
/b
On Jun 16, 2009, at 11:02 PM, Edmar Cruz wrote:
How can sofia profile can call ACL?
Can you give me an example?
Like this?
I put this on external profile
/
/
Brian West-3 wrote:
Now you have to tell the sofia profile to use that ACL
/b
On Jun 16, 2009, at 10:03 PM, Edmar Cruz wrote:
How can i turn off authentication? This is my acl.conf.xml on
192.168.0.105
list name=fsb default=deny
node type=allow cidr=192.168.0.104/32/
/list
On 192.168.0.4
list name=fsa default=deny
node type=allow cidr=192.168.0.105/32/
/list
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Re: [Freeswitch-users] Zoiper reject freeswitch calls
What's wrong of the contact string? 639(snom) works but 637(zoiper) doesn't. user sip:6...@192.168.1.27:5070;rinstance=e1e47e9a22f3e450;transport=UDP;fs_nat=yes;fs_path=sip%3A637%40192.168.1.27%3A5070%3Brinstance%3De1e47e9a22f3e450%3Btransport%3DUDP seven sip:6...@192.168.1.21:2051;line=298293g2;fs_nat=yes;fs_path=sip%3A639%40192.168.1.21%3A2051%3Bline%3D298293g2 On Jun 16, 2009, at 8:43 PM, Brian West wrote: Ok i'll have to se what I can do about reproducing this issue now that I have more info on how its happening. /b On Jun 16, 2009, at 7:40 AM, dujinfang wrote: Almost caught you on IRC Mike. Our server is in a NAT'd network and all agents registered in the same LAN. I can remotely register by using the public IP and the contact string is right. Call-ID:ZTZhMGJkZTE0NzNjZTlmZTkxYmU5NWRlZTU1MzJlYTE. User: 6...@192.168.1.16 Contact:user sip:6...@210.73.8.180:5090;rinstance=9fc589bbc407518f Agent: Zoiper rev.1809 So it's like only happens on our LAN and where there's a fs_path present. Just curious, why agents registered on a local LAN has param fs_nat=yes; (default internal profile, port 5060) ? Seems our time doesn't match, I'm generally available in office 9:00AM-6:00PM GMT+0800, so will try to catch you tomorrow. Thank you. ___ Freeswitch-users mailing list 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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
