November 2011 - freeswitch-dev@lists.freeswitch.org

[Freeswitch-dev] 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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[Freeswitch-dev] ignore me
by Michael Collins 10 Nov '11

10 Nov '11

test test

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[Freeswitch-dev] 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-dev] 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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[Freeswitch-dev] 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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[Freeswitch-dev] 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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[Freeswitch-dev] FreeSwitch date and time conditions
by Max Machula 03 Nov '11

03 Nov '11

Hello, Hope someone can help me. I have a strange problem with date and time on my FreeSwitch. I have dialplan: <extension name="WW26"> <condition field="destination_number" expression="26"/> <condition field="year" expression="2011"> <action application="bridge" data="USER/26@$${domain}"/> ... skip ... </condition> </extension> Server time: # date Thu Nov 3 21:34:22 GMT 2011 However FreeSwitch can not confirm the year 2011. Log file: Dialplan: sofia/internal/07711567890(a)xx.xx.xx.xx parsing [public->WW26] continue=true Dialplan: sofia/internal/07711567890(a)xx.xx.xx.xxRegex (PASS) [WW26] destination_number(26) =~ /26/ break=on-false Dialplan: sofia/internal/07711567890(a)xx.xx.xx.xxRegex (FAIL) [WW26] year() =~ /2011/ break=on-false I get the same problem if I change field year to wday which set to 1-6 for example now. Any ideas would be much appreciated. Thanks, Max

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[Freeswitch-dev] Send Event/User Channel Variable from FreeSWITCH to RTMP client
by RaviRaj Mulasa 03 Nov '11

03 Nov '11

Hi Please let us know how to send an Event/ User Channel Variable from FreeSWITCH to a RTMP client. Thanks RaviRaj.

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[Freeswitch-dev] FreeSWITCH Conference Call Today - New Dialplan Feature, SIP 101
by Michael Collins 02 Nov '11

02 Nov '11

Hello all! In our conference call today we are going to discuss a cool new dialplan feature (better logical OR processing) and then Ken Rice and I will be discussing some basic SIP knowledge to help get everyone up to speed when helping newbies. Talk to you all soon! -Michael

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