- FreeSWITCH-dev - FreeSWITCH Mailing Lists

version numbers
by Dalei Liu 12 Nov '11

12 Nov '11

Hi, I've been using freeswitch for a while and really like the stability and sip compatibility. While the code is excellent, I think the version management could be better. >From the git repository, the last tag is 1.0.6 dated a year and half ago, and there was a 1.0.7 announcement but no tag anywhere. As described in the installation guide git repository is strongly recommended but the problem is, it's a changing target. And using SHA1 is really not helpful to the users. Is it possible to define a formal version numbering system and regular release procedure? For example, if we think the current master is stable enough has the features we need and no major issue, tag it as 1.1-rc1, release it and let the users to test it under all different conditions. And then tag 1.1-rc2, 1.1-rc3, etc until all the the minor issues are either fixed or moved to next minor release v1.2. At that moment, tag and release 1.1 and create a branch b1.1 on master. Later, only the bug fixes go into the branch and it can be tagged as v1.1.1, then 1.1.2, etc. All the new features or enhancements will sit in master and be included in the next minor release v1.2, which will follow the same procedure. In this way, the users will have a fixed target to test and report issues and the developers will have a way to reproduce the same problem. And it will help the users and developers from outside understand the development progress. I believe it will also help integrate freeswitch into mainstream distros, like debian and fedora. My dream is one day, I could just type apt-get install freeswitch from the fresh debian installation. If you think it worths the time, I could contribute some time on it. Best regards, Dalei

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Re: [Freeswitch-dev] Cepstral locks up
by Tom Parrott 11 Nov '11

11 Nov '11

Hi, I am happy to, although I'm sure it can be improved as this is my first lua script. Although its in production now and working like a charm. :) First, I created a phrase macro for requesting TTS, and then call the LUA script from there:  <macro name="tts"> <input pattern="(.*)"> <match> <action function="execute" data="lua(tts_cache.lua '$1')"/> <action function="play-file" data="${tts_file}" /> </match> </input> </macro> Then this is the tts_cache.lua script: -- This script generates a wav file of the sentence passed in to it. -- It uses the Cepstral swift command to perform text-to-speech conversion. -- If the wav file already exists for this sentence, then it is not generated. -- Author: Thomas Parrott <thomas.parrott(a)infinity-tracking.com> api = freeswitch.API(); msg = argv[1]; msgMd5 = api:execute( "md5", msg ); filename = '/var/lib/tts_cache/' .. msgMd5 .. '.wav'; cmd = '/opt/swift/bin/swift'; -- Set a channel variable so that we know which file to play back. session:setVariable( 'tts_file', filename ); -- Check whether the file already exists. file, errMsg = io.open( filename, "r" ) if not file then api:execute( 'system', cmd .. ' -o "' .. filename .. '" "' .. msg .. '"' ); end If I can have a branch into contrib I'll post it up there. Thanks Tom > On 11/11/2011 01:00 AM, Tom Parrott wrote: > [snip] >> Let me know if you would like to see the script. > > Good to see you found a solution. If you don't mind sharing your script > with the Community perhaps you could ask one of the developers to put it > somewhere in contrib/ or give you access to your own branch in contrib/. > > Regards, > Patrick > > >

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Cepstral locks up
by Tom Parrott 11 Nov '11

11 Nov '11

Hi, I am using mod_cepstral to perform call announce features. On one of our servers the API call to run a TTS blocked, and stopped the call from proceeding until a 20 second delay had occurred. Additionally, when I ran /opt/swift/bin/swift -o test.wav "Hello World" The command would also block for 20 seconds, and not generate any audio. I ran reload mod_cepstra, to no avail. Eventually I restarted Freeswitch and it fixed the problem. Do you know what could cause this? I have a license purchase from Cepstral. Thanks Tom

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Testing 1 2 3
by Ken Rice 10 Nov '11

10 Nov '11

M Collins says ³TAP TAP TAP, Is this thing on??² K

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ignore me
by Michael Collins 10 Nov '11

10 Nov '11

test test

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Asterisk / FreeSWITCH Developer required..
by Robert Kirkby 10 Nov '11

10 Nov '11

Hi We are looking for a developer / expert for an ongoing project. The project would include development as well as consultancy on support and hosting options. We are looking for a telephone solution with the features of a conference bridge, a simple outbound dialer with a large emphasis on call recording. The candidate will need a good knowledge of hosting Asterisk / FreeSWITCH , specifically in an environment where call quality is important, so knowledge of data centres with access to ISDN lines, not just VOIP. A full specification of the project will be provided after initial contact. Please email me on: rob.kirkby(a)gmail.com Thanks Rob Kirkby

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FreeSWITCH Conference Call Today
by Michael Collins 09 Nov '11

09 Nov '11

Hello FreeSWITCHers! Here's today's agenda: http://wiki.freeswitch.org/wiki/FS_weekly_2011_11_09 We have a few miscellaneous items to discuss and I have some updates. Also, I'm hoping to continue the discussion on SIP 101 that we started last week. I think it would be a good time to go over the Contact: header and what it's for. That will set the ground work for what the NDLB-connectile-dysfunction parameter does. Talk to you soon, -Michael

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subscribe to a gateway
by Seven Du 05 Nov '11

05 Nov '11

Hi, Can freeswitch as a client subscribe to a gateway(maybe another freeswitch) to get notifications? If not, is there any plan to do it or highlight on how hard to implement and where should I start from code? Thanks. -- About: http://about.me/dujinfang Blog: http://www.dujinfang.com Proj: http://www.freeswitch.org.cn Sent with Sparrow (http://www.sparrowmailapp.com)

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How to display callee_id_name on the caller's phone
by Han Xin 04 Nov '11

04 Nov '11

Hi all, There is a bridged call, the caller_id_name of leg-A can be displayed correctly on both FS console and callee's phone, but the callee_id_name was always displayed as "Outbound Call"... Some variables during the call are dumped as follows: uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,application,application_data,dialplan,context,read_codec,read_rate,read_bit_rate,write_codec,write_rate,write_bit_rate,secure,hostname,presence_id,presence_data,callstate,callee_name,callee_num,callee_direction,call_uuid,sent_callee_name,sent_callee_num 8e4a992c-d82f-4919-aee4-6ac0190784ce,inbound,2011-11-02 14:29:15,1320240555,sofia/internal/1000@asterisk,CS_EXECUTE,Bob 1000 (FS),1000,137.226.198.252,1001,bridge,user/1001@137.226.198.143,XML,default,G722,16000,64000,G722,16000,64000,,asterisk,1000 @asterisk,,ACTIVE,Outbound Call,1001,SEND,8e4a992c-d82f-4919-aee4-6ac0190784ce,Outbound Call,1001 be08b9b5-ccd8-44e0-bbd8-4ca93841f930,outbound,2011-11-02 14:29:15,1320240555,sofia/internal/sip:1001@137.226.198.252 :5060,CS_EXCHANGE_MEDIA,Bob 1000,1000,137.226.198.252,1001,,,XML,default,G722,16000,64000,G722,16000,64000,,asterisk,1001(a)137.226.198.143,,ACTIVE,Outbound Call,1001,SEND,8e4a992c-d82f-4919-aee4-6ac0190784ce,Bob 1000 (FS),1000 And the configuration of user "Alice 1001"(in file 1001.xml) is like this: (Bob 1000 is also the same kind of config) <include> <user id="1001"> </params> <variables> <variable name="toll_allow" value="domestic,international,local"/> <variable name="accountcode" value="1001"/> <variable name="user_context" value="default"/> <variable name="effective_caller_id_name" value="Alice 1001"/> <variable name="effective_caller_id_number" value="1001"/> <variable name="outbound_caller_id_name" value="Alice"/> <variable name="outbound_caller_id_number" value="1001"/> <variable name="effective_callee_id_name" value="Alice 1001"/> <variable name="effective_callee_id_number" value="1001"/> <variable name="callgroup" value="techsupport"/> </variables> </user> </include> why the "effective_callee_id_name" does not work? Is there something I have not done in order to display the real callee_id_name? Thanks in advance. Best Regards, Han --

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Re: [Freeswitch-dev] Nortel 1535 drops connection to FS
by Oleg Khovayko 04 Nov '11

04 Nov '11

Anthony, I found problem source. I think, it will be valuable information for our community. Also, maybe will need to fix FS according this info. 1. About "FS does not receive ACK" - you was right. Really, FS was not received ACKs, and breaks connection. 2. Currently, FS runs behind NAT router, with option "-nonat" and static routing. Also, there is firewall on the FS server, which allows connection to port 5060 only from some selected networks. Server has static IP=192.168.1.5, WAN IP=173.79.237.11, Phone connected to LAN, IP=192.168.1.158 (DHCP). 3. When I (and Stan G.) investigated this problem, we run tcpdump _inside_ the phone, and found -- phone sends ACKs to WAN ROUTERs port (173.79.237.11), when must send to FS server, 192.168.1.5 {{{ tcpdump -A -vvv port 5060 Via: SIP/2 20:36:27.790809 IP (tos 0xa0, ttl 64, id 28499, offset 0, flags [none], length: 699) lgvp.5060 > Wireless_Broadband_Router.5060: UDP, length: 671 E...oS..@..8.....O.p......%GACK sip:5000@173.79.237.11 20:36:27.809815 IP (tos 0xa0, ttl 64, id 28500, offset 0, flags [none], length: 699) lgvp.5060 > Wireless_Broadband_Router.5060: UDP, length: 671 E...oT..@..7.....O.p......%GACK sip:5000@173.79.237.11 20:36:27.815188 IP (tos 0xa0, ttl 64, id 28501, offset 0, flags [none], length: 699) lgvp.5060 > Wireless_Broadband_Router.5060: UDP, length: 671 E...oU..@..6.....O.p......%GACK sip:5000@173.79.237.11 20:36:29.179099 IP (tos 0x0, ttl 64, id 43922, offset 0, flags [none], length: 1174) 192.168.1.5.5060 > lgvp.5060: UDP, length: 1146 E.......@.F.................SIP/2.0 200 OK }} When router sends UDP packets to FS box, they receiving from Router's address 173.79.237.11, and rejected by IP firewall. I allowed my own WAN subnet 173.79.0.0/16 with firewall, and thereafter phone works OK. Following - sample of the FS log, when connection works OK, with ACKs. http://olegh.ath.cx/freeswitch.log-1535-2 <http://olegh.ath.cx/freeswitch.log-1535-1> But, question: How can I enforce phone works directly with FS, without loopback in the router? I assume, phone sends request to WAN address instead of direct FS, because FS sends to LAN client contact field, contains WAN address in the SIP 200 message: Contact: <sip:5000@173.79.237.112:5060;transport=udp> Of course, client sends replies back to this WAN address... How to fix this issue, for FS send to local clients only something like Contact: <sip:5000@192.168.1.5:5060;transport=udp> Thanks, Oleg Anthony Minessale wrote: > you are not getting an ACK back from the phone after the 200OK > investigate the sip traffic and look for something like SIP ALG or > other things that may cause the ACK to not make it back to FS. > > > 2011/11/3 Oleg Khovayko <khovayko(a)gmail.com <mailto:khovayko@gmail.com>> > > Anthony, > > > Thank you very much for assistance. I produced log-file, contains > siptrace, you can get it: > > http://olegh.ath.cx/freeswitch.log-1535-1 > > In thins test, I called from numbed 1000 to number 5000. > > > Oleg > > > > > > > > > Anthony Minessale wrote: >> snippets of logs rarely are helpful, try complete debug logs >> including sip trace >> >> >> On Wed, Nov 2, 2011 at 4:16 PM, Oleg Khovayko <khovayko(a)gmail.com >> <mailto:khovayko@gmail.com>> wrote: >> >> Hi, >> >> I am using FS for couple years, it was works OK. >> And, approx 1 year ago, I tested with FreeSWITCH phone Nortel >> 1535. >> It worked perfect. >> >> Thereafter, I couple times upgraded FS, and right now, I run >> version >> from GIT, dated 11-10-29. OS: FreeBSD 7.2, 32 bits. >> >> I again connected Nortel 1535, and this time see, connection >> suddenly >> broken during call to any "robot": >> FS #5000, FS #4000, CallCentric #17771234567 >> >> Every time, before disconnect, I see chain of log messages like: >> >> [DEBUG] switch_core_session.c:872 Send signal >> sofia/internal/1000(a)olegh.ath.cx <mailto:1000@olegh.ath.cx> >> [BREAK] >> and HANGUP thereafter. See following samples from the log: >> >> Another phones, like VM1188T or PAP2, still works OK. >> About Nortel: I tested 3 phones, with two firmwares (2.50 & >> 2.76) -- >> problem same on both. >> >> Can you suggest me, how to fix this issue? >> >> Thanks, >> >> Oleg >> >> >> >> CallCentric: 7771234567 >> >> 2011-11-02 16:39:20.901219 [DEBUG] sofia_reg.c:1984 Changing >> expire time >> to 103 by request of proxy sip:callcentric.com >> <http://callcentric.com> >> 2011-11-02 16:39:31.108503 [DEBUG] switch_core_session.c:872 >> Send signal >> sofia/internal/1000(a)olegh.ath.cx <mailto:1000@olegh.ath.cx> >> [BREAK] >> 2011-11-02 16:39:31.108503 [DEBUG] switch_core_session.c:872 >> Send signal >> sofia/internal/1000(a)olegh.ath.cx <mailto:1000@olegh.ath.cx> >> [BREAK] >> 2011-11-02 16:39:31.108503 [DEBUG] switch_core_session.c:872 >> Send signal >> sofia/internal/1000(a)olegh.ath.cx <mailto:1000@olegh.ath.cx> >> [BREAK] >> 2011-11-02 16:39:31.108503 [DEBUG] switch_core_session.c:872 >> Send signal >> sofia/internal/1000(a)olegh.ath.cx <mailto:1000@olegh.ath.cx> >> [BREAK] >> 2011-11-02 16:39:31.130669 [DEBUG] switch_core_session.c:872 >> Send signal >> sofia/internal/1000(a)olegh.ath.cx <mailto:1000@olegh.ath.cx> >> [BREAK] >> 2011-11-02 16:39:31.130669 [DEBUG] sofia.c:5283 Channel >> sofia/internal/1000(a)olegh.ath.cx <mailto:1000@olegh.ath.cx> >> entering state [terminating][0] >> 2011-11-02 16:39:31.130669 [DEBUG] switch_channel.c:2804 >> (sofia/internal/1000(a)olegh.ath.cx <mailto:1000@olegh.ath.cx>) >> Callstate Change ACTIVE -> HANGUP >> 2011-11-02 16:39:31.130669 [NOTICE] sofia.c:6039 Hangup >> sofia/internal/1000(a)olegh.ath.cx <mailto:1000@olegh.ath.cx> >> [CS_EXECUTE] [NORMAL_UNSPECIFIED] >> 2011-11-02 16:39:31.130669 [DEBUG] switch_channel.c:2820 Send >> signal >> sofia/internal/1000(a)olegh.ath.cx <mailto:1000@olegh.ath.cx> >> [KILL] >> 2011-11-02 16:39:31.130669 [DEBUG] switch_core_session.c:1177 >> Send >> signal sofia/internal/1000(a)olegh.ath.cx >> <mailto:1000@olegh.ath.cx> [BREAK] >> >> >> FS: 5000 >> >> 2011-11-02 17:03:08.207852 [DEBUG] switch_ivr_play_say.c:1672 >> done >> playing file >> /usr/local/freeswitch/sounds/ru/RU/elena/ivr/ivr-to_do_a_freeswitch_echo_test.wav >> 2011-11-02 17:03:08.459809 [DEBUG] switch_ivr_play_say.c:244 >> Handle >> play-file:[ivr/ivr-please.wav] (ru:ru) >> 2011-11-02 17:03:08.459809 [DEBUG] switch_ivr_play_say.c:1302 >> Codec >> Activated L16@8000hz 1 channels 20ms >> 2011-11-02 17:03:09.257683 [DEBUG] switch_core_session.c:872 >> Send signal >> sofia/internal/1000(a)olegh.ath.cx <mailto:1000@olegh.ath.cx> >> [BREAK] >> 2011-11-02 17:03:09.257683 [DEBUG] switch_core_session.c:872 >> Send signal >> sofia/internal/1000(a)olegh.ath.cx <mailto:1000@olegh.ath.cx> >> [BREAK] >> 2011-11-02 17:03:09.257683 [DEBUG] switch_core_session.c:872 >> Send signal >> sofia/internal/1000(a)olegh.ath.cx <mailto:1000@olegh.ath.cx> >> [BREAK] >> 2011-11-02 17:03:09.280490 [DEBUG] switch_core_session.c:872 >> Send signal >> sofia/internal/1000(a)olegh.ath.cx <mailto:1000@olegh.ath.cx> >> [BREAK] >> 2011-11-02 17:03:09.280490 [DEBUG] sofia.c:5283 Channel >> sofia/internal/1000(a)olegh.ath.cx <mailto:1000@olegh.ath.cx> >> entering state [terminating][0] >> 2011-11-02 17:03:09.280490 [DEBUG] switch_channel.c:2804 >> (sofia/internal/1000(a)olegh.ath.cx <mailto:1000@olegh.ath.cx>) >> Callstate Change ACTIVE -> HANGUP >> 2011-11-02 17:03:09.280490 [NOTICE] sofia.c:6039 Hangup >> sofia/internal/1000(a)olegh.ath.cx <mailto:1000@olegh.ath.cx> >> [CS_EXECUTE] [NORMAL_UNSPECIFIED] >> 2011-11-02 17:03:09.280490 [DEBUG] switch_channel.c:2820 Send >> signal >> sofia/internal/1000(a)olegh.ath.cx <mailto:1000@olegh.ath.cx> >> [KILL] >> 2011-11-02 17:03:09.280490 [DEBUG] switch_core_session.c:1177 >> Send >> signal sofia/internal/1000(a)olegh.ath.cx >> <mailto:1000@olegh.ath.cx> [BREAK] >> 2011-11-02 17:03:09.280490 [DEBUG] switch_ivr_play_say.c:1672 >> done >> playing file >> /usr/local/freeswitch/sounds/ru/RU/elena/ivr/ivr-please.wav >> 2011-11-02 17:03:09.532638 [DEBUG] switch_ivr_menu.c:348 >> waiting for 4/4 >> digits t/o 2000 >> 2011-11-02 17:03:09.532638 [DEBUG] switch_ivr_menu.c:395 >> digits '' >> 2011-11-02 17:03:09.532638 [DEBUG] switch_ivr_menu.c:585 IVR menu >> 'demo_ivr' no input detected >> 2011-11-02 17:03:10.587476 [DEBUG] switch_ivr_menu.c:599 >> exit-sound >> 'voicemail/vm-goodbye.wav' >> 2011-11-02 17:03:10.587476 [DEBUG] switch_core_session.c:2269 >> sofia/internal/1000(a)olegh.ath.cx <mailto:1000@olegh.ath.cx> >> skip receive message >> [APPLICATION_EXEC_COMPLETE] (channel is hungup already) >> >> >> FS: voicemail >> >> 2011-11-02 16:51:50.217856 [DEBUG] switch_ivr_play_say.c:67 >> No language >> specified - Using [ru] >> 2011-11-02 16:51:50.307721 [DEBUG] switch_ivr_play_say.c:244 >> Handle >> say:[1290554584] (ru:ru) >> 2011-11-02 16:51:50.307721 [INFO] mod_say_ru.c:721 ru_say!!! >> 1290554584! say_opt.gender=0 say_opt.cases=0 >> 2011-11-02 16:51:50.307721 [DEBUG] switch_ivr_play_say.c:1302 >> Codec >> Activated L16@8000hz 1 channels 20ms >> 2011-11-02 16:51:50.969016 [DEBUG] switch_core_session.c:872 >> Send signal >> sofia/internal/1000(a)olegh.ath.cx <mailto:1000@olegh.ath.cx> >> [BREAK] >> 2011-11-02 16:51:50.969016 [DEBUG] switch_core_session.c:872 >> Send signal >> sofia/internal/1000(a)olegh.ath.cx <mailto:1000@olegh.ath.cx> >> [BREAK] >> 2011-11-02 16:51:50.969016 [DEBUG] switch_core_session.c:872 >> Send signal >> sofia/internal/1000(a)olegh.ath.cx <mailto:1000@olegh.ath.cx> >> [BREAK] >> 2011-11-02 16:51:50.969016 [DEBUG] switch_core_session.c:872 >> Send signal >> sofia/internal/1000(a)olegh.ath.cx <mailto:1000@olegh.ath.cx> >> [BREAK] >> 2011-11-02 16:51:50.969016 [DEBUG] switch_core_session.c:872 >> Send signal >> sofia/internal/1000(a)olegh.ath.cx <mailto:1000@olegh.ath.cx> >> [BREAK] >> 2011-11-02 16:51:50.990445 [DEBUG] sofia.c:5283 Channel >> sofia/internal/1000(a)olegh.ath.cx <mailto:1000@olegh.ath.cx> >> entering state [terminating][0] >> 2011-11-02 16:51:50.990445 [DEBUG] switch_channel.c:2804 >> (sofia/internal/1000(a)olegh.ath.cx <mailto:1000@olegh.ath.cx>) >> Callstate Change ACTIVE -> HANGUP >> 2011-11-02 16:51:50.990445 [NOTICE] sofia.c:6039 Hangup >> sofia/internal/1000(a)olegh.ath.cx <mailto:1000@olegh.ath.cx> >> [CS_EXECUTE] [NORMAL_UNSPECIFIED] >> 2011-11-02 16:51:50.990445 [DEBUG] switch_channel.c:2820 Send >> signal >> sofia/internal/1000(a)olegh.ath.cx <mailto:1000@olegh.ath.cx> >> [KILL] >> 2011-11-02 16:51:50.990445 [DEBUG] switch_core_session.c:1177 >> Send >> signal sofia/internal/1000(a)olegh.ath.cx >> <mailto:1000@olegh.ath.cx> [BREAK] >> >> >> _______________________________________________ >> Join us at ClueCon 2011, Aug 9-11, Chicago >> http://www.cluecon.com 877-7-4ACLUE >> >> FreeSWITCH-dev mailing list >> FreeSWITCH-dev(a)lists.freeswitch.org >> <mailto:FreeSWITCH-dev@lists.freeswitch.org> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev >> 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_minessale@hotmail.com >> <mailto:MSN%3Aanthony_minessale@hotmail.com> >> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com >> <mailto:PAYPAL%3Aanthony.minessale@gmail.com> >> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888@conference.freeswitch.org >> <mailto:sip%3A888@conference.freeswitch.org> >> googletalk:conf+888@conference.freeswitch.org >> <mailto:googletalk%3Aconf%2B888@conference.freeswitch.org> >> pstn:+19193869900 <tel:%2B19193869900> >> >> >> _______________________________________________ >> Join us at ClueCon 2011, Aug 9-11, Chicago >> http://www.cluecon.com 877-7-4ACLUE >> >> FreeSWITCH-dev mailing list >> FreeSWITCH-dev(a)lists.freeswitch.org <mailto:FreeSWITCH-dev@lists.freeswitch.org> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev >> 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_minessale@hotmail.com > <mailto:MSN%3Aanthony_minessale@hotmail.com> > GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com > <mailto:PAYPAL%3Aanthony.minessale@gmail.com> > IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch > > FreeSWITCH Developer Conference > sip:888@conference.freeswitch.org > <mailto:sip%3A888@conference.freeswitch.org> > googletalk:conf+888@conference.freeswitch.org > <mailto:googletalk%3Aconf%2B888@conference.freeswitch.org> > pstn:+19193869900

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