[Freeswitch-users] Digium TE220 wiki
Hi all, I've created a wiki page, which contains the example configuration for making Digium TE220 to work. I request you people to check this, and give feedbacks. http://wiki.freeswitch.org/wiki/Configuration_OpenZap-DigiumTE220-Example ___ 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] Qustion about INFO messages after Connect/Answer
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Hash: SHA1
Hello Anthony,
On 19.10.2009 22:07, Anthony Minessale wrote:
please update and test trunk
1) I changed the core to remove the excess data by default in your scenario
2) I added variables you can use to control it
origination_callee_id_name origination_callee_id_number which belong in
{} in the dial string eg {origination_callee_id_number=1234}openzap/1/a/1234
updated and testet for SIP calls. origination_callee_id_number=1234
works within the dial string and via using export application but not
via set application. Didn't test it for openzap, yet but I guess it
will work, too.
What left is that unknown thing on callee's display (SIP phones only I
guess) ... I would suggest to make sending unknown INFO message
optional, but to be honest, I have no idea for what it was invented, so
maybe my suggestion is nonsens.
regards
Helmut
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[Freeswitch-users] Stereo Support
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hello, I wonder if there is real stereo support planned for FreeSWITCH in terms of streaming music/video-audio to desktop? I ask, because I heared that celt codec is supporting stereo. regards helmut -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (MingW32) iD8DBQFK3Whc4tZeNddg3dwRAkdEAKCLBSV4H8q5JN4NZ39fgSZPxr5PCgCgmpTq PAu8qozLP5FaVKIwfCXVqjU= =Zseo -END PGP SIGNATURE- ___ 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] Is anyone running Ubuntu 8.04/Hardy?
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi Mark, I confirm that it works fine. I use it for two years now and there were no problems compiling or running FS on ubuntu 8.04 caused by the OS. This is my system: lsb_release -a No LSB modules are available. Distributor ID: Ubuntu Description:Ubuntu 8.04.3 LTS Release:8.04 Codename: hardy uname -a Linux ippbx-prod-node0 2.6.24-22-server #1 SMP Mon Nov 24 19:14:19 UTC 2008 i686 GNU/Linux On 19.10.2009 19:35, Mark Sobkow wrote: Everyone I've emailed with on the dev list is running the current release of Ubuntu, not 8.04/Hardy. We're having some problems with loading mod_sofia on Hardy, so unless I get some progress/help by end of day Wednesday we're going to have to rebuild our development server with the most recent release of Ubuntu on Thursday and Friday. regards helmut -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (MingW32) iD8DBQFK3Wkz4tZeNddg3dwRAntOAJ97Gc67k8iEU/a6vhbb+YZMYoq2FQCgudHX 3b5Ya5ueoiEY7O4QIrni73U= =95L5 -END PGP SIGNATURE- ___ 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: mod_opal - call charged before H.225 connect
TC TCcall flow is SIP_user = FS = H323_endpoint is failing .. coredumped TChttp://pastebin.freeswitch.org/10703 i fix some bugs now, ftp://srv.icf.org.ru/pub/soft/f/freeswitch/mod_h323/mod_h323.tar.bz2 this is updated version, try it, if you experience no audio try enable rtp proxy in you sip profile. Hi, there are several issues... lets start with top 4 :) 1. I'm still stuck with no audio: I have this parameter in the sip profile set: param name=inbound-proxy-media value=true/ ...tried with both with slow start and fast start... any idea ? pls check: http://pastebin.freeswitch.org/10771 2. outgoing calls still failing in coredumps: what is your dialplan ? ... how do you call bridge application? 2009-10-20 10:08:18.424083 [DEBUG] h323caps.cxx:3252 Found capability: UserInput/PointDevice 14 2009-10-20 10:08:18.424083 [DEBUG] h323caps.cxx:3248 FindCapability: 15 2009-10-20 10:08:18.426608 [DEBUG] osutils.cxx:60 Created external thread 0xb6eb60a0 for id 3048876944 2009-10-20 10:08:18.426608 [DEBUG] osutils.cxx:65 Destroyed external thread 0xb6ebafa8 for id 3048876944 2009-10-20 10:08:18.426608 [DEBUG] tlibthrd.cxx:406 Destroyed thread 0xb6ebafa8 PExternalThread:0xb5ba2b90(id = b5ba2b90) 2009-10-20 10:08:18.426608 [DEBUG] h323caps.cxx:3252 Found capability: UserInput/Modal 15 2009-10-20 10:08:18.426608 [DEBUG] osutils.cxx:880 MONITOR: timers=0, expiries=0 2009-10-20 10:08:18.426608 [DEBUG] osutils.cxx:880 MONITOR: timers=0, expiries=0 2009-10-20 10:08:18.426608 [DEBUG] osutils.cxx:60 Created external thread 0xb6eba910 for id 3048876944 2009-10-20 10:08:18.426608 [DEBUG] h4601.cxx:1725 Endpoint Attached Segmentation fault (core dumped) tculj...@subzero:~/freeswitch-trunk$ pls check: http://pastebin.freeswitch.org/10769 3. when you hangup from SIP side, the call is not released end-to-end (the H323 endpoint doesn't get any releaseComplete message) 2009-10-20 10:10:51.264527 [DEBUG] osutils.cxx:880 MONITOR: timers=2, expiries=3 2009-10-20 10:10:51.264527 [DEBUG] h323neg.cxx:432 Received MasterSlaveDeterminationAck: state=Incoming 2009-10-20 10:10:51.264527 [DEBUG] osutils.cxx:880 MONITOR: timers=1, expiries=3 2009-10-20 10:10:51.264527 [DEBUG] osutils.cxx:880 MONITOR: timers=2, expiries=4 2009-10-20 10:10:51.264527 [DEBUG] h323.cxx:4138 InternalEstablishedConnectionCheck: connectionState=EstablishedConnection fastStartState=FastStartAcknowledged 2009-10-20 10:10:51.264527 [DEBUG] h323caps.cxx:3264 FindCapability: T.120 2009-10-20 10:10:51.264527 [DEBUG] h323.cxx:4138 InternalEstablishedConnectionCheck: connectionState=EstablishedConnection fastStartState=FastStartAcknowledged 2009-10-20 10:10:51.264527 [DEBUG] tlibthrd.cxx:1023 PThread::PXBlockOnIO(45,0) 2009-10-20 10:10:54.405479 [NOTICE] sofia.c:328 Hangup sofia/internal/ sip:1...@10.4.62.89 sip%3a1...@10.4.62.89 [CS_CONSUME_MEDIA] [NORMAL_CLEARING] 2009-10-20 10:10:54.405479 [DEBUG] switch_channel.c:1726 Send signal sofia/internal/sip:1...@10.4.62.89 sip%3a1...@10.4.62.89 [KILL] 2009-10-20 10:10:54.405479 [DEBUG] switch_core_session.c:932 Send signal sofia/internal/sip:1...@10.4.62.89 sip%3a1...@10.4.62.89 [BREAK] 2009-10-20 10:10:54.405479 [DEBUG] switch_core_state_machine.c:437 thread mismatch skipping state handler. 2009-10-20 10:10:54.405479 [DEBUG] switch_core_state_machine.c:306 (sofia/internal/sip:1...@10.4.62.89 sip%3a1...@10.4.62.89) Running State Change CS_HANGUP 2009-10-20 10:10:54.406530 [DEBUG] switch_core_state_machine.c:464 (sofia/internal/sip:1...@10.4.62.89 sip%3a1...@10.4.62.89) State HANGUP 2009-10-20 10:10:54.406530 [DEBUG] mod_sofia.c:338 Channel sofia/internal/ sip:1...@10.4.62.89 sip%3a1...@10.4.62.89 hanging up, cause: NORMAL_CLEARING 2009-10-20 10:10:54.406530 [DEBUG] switch_core_state_machine.c:46 sofia/internal/sip:1...@10.4.62.89 sip%3a1...@10.4.62.89 Standard HANGUP, cause: NORMAL_CLEARING pls check: http://pastebin.freeswitch.org/10771 4. sometimes when i shutdown FS i get coredimps - from my experience it looks like you don't wait for a FS thread to finish when you exit... 2009-10-20 10:05:59.493306 [CONSOLE] switch_event.c:508 Stopping queue thread 2 2009-10-20 10:05:59.493339 [CONSOLE] switch_core.c:1693 Finalizing Shutdown. 2009-10-20 10:05:59.493379 [CONSOLE] switch_log.c:310 Logger Ended. 2009-10-20 10:05:59.494472 [CONSOLE] switch_core_memory.c:567 Stopping memory pool queue. Segmentation fault (core dumped) tculj...@subzero:~/freeswitch-trunk$ tculj...@subzero:~/freeswitch-trunk$ Please advice your FS/mod_h323.conf.xml settings... Tihomir. C уважением With Best Regards Георгиевский Юрий.Georgiewskiy Yuriy +7 4872 711666+7 4872 711666 факс +7 4872 711143 fax +7 4872 711143 Компания ООО Ай Ти Сервис IT Service Ltd http://nkoort.ru http://nkoort.ru JID: ghh...@jabber.tula-ix.net.ru JID: ghh...@jabber.tula-ix.net.ru YG129-RIPEYG129-RIPE
[Freeswitch-users] Can not record session. Media not enabled on channel.
Hello, I am using the same set of extensions for testing the system during development, they include XLite, Cisco sip phone and several extensions that just play some audio file. Sometimes, very rarely, this message Can not record session. Media not enabled on channel. appears on FS console. Like I wrote before, I don't change any codec settings and always use the same set of devices/emulators. What can cause this message? Thanks, Maciej Aniserowicz -- View this message in context: http://n2.nabble.com/Can-not-record-session-Media-not-enabled-on-channel-tp3857858p3857858.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
Re: [Freeswitch-users] uuid_send_dtmf fails (was: conference call)
Yeah. A call to B and C eavesdrops call. I send dtmf for C to talk with B, but B can't hear C. Here, what I had done in details: [r...@centos4-4-vm ~]# telnet localhost 8021 Trying 127.0.0.1... Connected to localhost.localdomain (127.0.0.1). Escape character is '^]'. Content-Type: auth/request auth ClueCon Content-Type: command/reply Reply-Text: +OK accepted api originate user/1...@master.agent.starpoundtech.net park() Content-Type: api/response Content-Length: 41 +OK bba3b45a-4cc1-48af-a15d-1052d5f11371 SendMsg bba3b45a-4cc1-48af-a15d-1052d5f11371 call-command: execute execute-app-name: eavesdrop execute-app-arg: cd99f999-9b47-457e-8439-1d366e015b8c Content-Type: command/reply Reply-Text: +OK Here I had started to hear A and B. Here what I saw in FS log: 2009-10-18 03:22:47 [DEBUG] switch_core_session.c:706 switch_core_session_queue_private_event() Send signal sofia/internal/sip:1...@172.26.10.64:5060;fs_nat=yes [BREAK] 2009-10-18 03:22:47 [DEBUG] switch_ivr.c:540 switch_ivr_parse_event() sofia/internal/sip:1...@172.26.10.64:5060;fs_nat=yes Command Execute eavesdrop(cd99f999-9b47-457e-8439-1d366e015b8c) 2009-10-18 03:22:47 [DEBUG] switch_core_media_bug.c:297 switch_core_media_bug_add() Attaching BUG to sofia/internal/1...@master.agent.starpoundtech.net 2009-10-18 03:22:47 [DEBUG] switch_core_io.c:234 switch_core_session_read_frame() sofia/internal/1...@master.agent.starpoundtech.net receive message [TRANSCODING_NECESSARY] 2009-10-18 03:22:47 [DEBUG] switch_core_io.c:234 switch_core_session_read_frame() sofia/internal/sip:1...@172.26.10.64:5060;fs_nat=yes receive message [TRANSCODING_NECESSARY] Then I run command: api uuid_send_dtmf bba3b45a-4cc1-48af-a15d-1052d5f11371 1 Content-Type: api/response Content-Length: 14 -ERR no reply Log: 2009-10-18 03:24:01 [DEBUG] switch_core_io.c:1190 switch_core_session_send_dtmf_string() sofia/internal/sip:1...@172.26.10.64:5060;fs_nat=yes send dtmf digit=1 ms=250 samples=2000 2009-10-18 03:24:01 [DEBUG] switch_rtp.c:1282 do_2833() Send start packet for [1] ts=2241760 dur=160/160/2000 seq=21346 2009-10-18 03:24:01 [DEBUG] switch_rtp.c:1221 do_2833() Send middle packet for [1] ts=2241760 dur=320/320/2000 seq=21347 2009-10-18 03:24:01 [DEBUG] switch_rtp.c:1221 do_2833() Send middle packet for [1] ts=2241760 dur=480/480/2000 seq=21348 2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send middle packet for [1] ts=2241760 dur=640/640/2000 seq=21349 2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send middle packet for [1] ts=2241760 dur=800/800/2000 seq=21350 2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send middle packet for [1] ts=2241760 dur=960/960/2000 seq=21351 2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send middle packet for [1] ts=2241760 dur=1120/1120/2000 seq=21352 2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send middle packet for [1] ts=2241760 dur=1280/1280/2000 seq=21353 2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send middle packet for [1] ts=2241760 dur=1440/1440/2000 seq=21354 2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send middle packet for [1] ts=2241760 dur=1600/1600/2000 seq=21355 2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send middle packet for [1] ts=2241760 dur=1760/1760/2000 seq=21356 2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send middle packet for [1] ts=2241760 dur=1920/1920/2000 seq=21357 2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send end packet for [1] ts=2241760 dur=2080/2080/2000 seq=21358 2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send end packet for [1] ts=2241760 dur=2080/2080/2000 seq=21359 2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send end packet for [1] ts=2241760 dur=2080/2080/2000 seq=21360 But both A and B couldn't hear me. Btw, after I had send dtmf 1 manually from my phone. B started to hear me. There was this record in log: 2009-10-18 03:47:55 [DEBUG] switch_rtp.c:1767 switch_rtp_dequeue_dtmf() RTP RECV DTMF 1:2240 _ From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Brian West Sent: Tuesday, October 20, 2009 4:54 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] uuid_send_dtmf fails (was: conference call) OK what makes you think it failed? The fact you don't hear it? /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] How to retain the B-number in an Call forward
Hello, I us a call forward on freeswitch to forward calls to my mobile phone. If now Freeswitch forwards a call, the number information presented to my mobile phone is that of the original caller, because that is what there is appearing in its display. What I want to see is that Freeswitch is calling me so I would know it is a business call and not an private call. Now I have no means to distinguishes between them. Kind regards, Durk de Beer ___ 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] mod_curl + mod_nibblebill
Hi, Consider below mob_nibblebill configuration, configuration name=nibblebill.conf description=Nibble Billing settings param name=db_username value=root/ param name=db_password value=password/ param name=db_dsn value=MySQL-freeswitch/ param name=db_table value=accounts/ param name=db_column_cash value=cash/ param name=db_column_account value=id/ param name=global_heartbeat value=60/ param name=lowbal_amt value=5/ param name=lowbal_action value=play ding/ param name=nobal_amt value=0/ param name=nobal_action value=hangup/ param name=percall_max_amt value=100/ param name=percall_action value=hangup/ /settings /configuration When i put this configuration in /usr/local/freeswitch/conf/autoload_configs, everything works perfect as expected. But when i call this configuration using mod_xml_curl, freeswitch does not loads mod_nibblebill configuration from it. I even debuged mod_xml_curl using console command, xml_curl debug_on and reload mod_nibblebill, I get exact same configuration from xml_curl but mod_nibblebill does not repect it and try to use default parameter values as mentioned below, 2009-10-20 16:58:23.396710 [ERR] switch_odbc.c:188 STATE: IM002 CODE 0 ERROR: [unixODBC][Driver Manager]Data source name not found, and no default driver specified 2009-10-20 16:58:23.396738 [CRIT] mod_nibblebill.c:221 Cannot connect to ODBC driver/database bandwidth.com (user: bandwidth.com / pass dev)! 2009-10-20 16:58:23.396800 [CONSOLE] switch_loadable_module.c:889 Successfully Loaded [mod_nibblebill] I am also loading various other configs from xml_curl without any problem, its only mod_nibblebill that is not honoring configuration provided by xml_curl. Please advise if i am doing anything wrong. -- Regards, Ahmed Munir ___ 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] mod_odbc_query share memory problem
Dear Sir, I'm using mod_odbc_query and mod_nibble_billing for my calling card solutoin. i found mod_odbc_query cant work with high load (200 calls concurrent) i got error [STATE: 53200 CODE 7 ERROR: [unixODBC]ERROR: out of shared memory; Error while executing the query ] in FS console How to fix it ? BG Dome 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] can't dial from IPv6 to IPv6
On 20/10/09 07:53 +0200, ineya ineya wrote: IP should be OK: [r...@franta /opt/freeswitch]# ifconfig eth1 eth1 Link encap:Ethernet HWaddr 00:4f:4e:62:ad:83 inet addr:10.80.62.40 Bcast:10.80.62.255 Mask:255.255.255.0 inet6 addr: 2000:2::1/32 Scope:Global This address caught my eye. Although valid, a /32 address assignment is atypical. Is that what you intended? Does assigning a /64 address on both ends make any difference? -- Dan White ___ 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_odbc_query share memory problem
Hi Dome, Have you tried increasing global_heartbeat to reduce the frequency of odbc calls? What is it currently set to? Rob On Oct 20, 2009, at 8:31 AM, Dome Charoenyost wrote: Dear Sir, I'm using mod_odbc_query and mod_nibble_billing for my calling card solutoin. i found mod_odbc_query cant work with high load (200 calls concurrent) i got error [STATE: 53200 CODE 7 ERROR: [unixODBC]ERROR: out of shared memory; Error while executing the query ] in FS console How to fix it ? BG Dome 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 ___ 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_odbc_query share memory problem
2009/10/20 Rob Forman rob4manh...@gmail.com: Hi Dome, Have you tried increasing global_heartbeat to reduce the frequency of odbc calls? What is it currently set to? Now 1 min Rob On Oct 20, 2009, at 8:31 AM, Dome Charoenyost wrote: Dear Sir, I'm using mod_odbc_query and mod_nibble_billing for my calling card solutoin. i found mod_odbc_query cant work with high load (200 calls concurrent) i got error [STATE: 53200 CODE 7 ERROR: [unixODBC]ERROR: out of shared memory; Error while executing the query ] in FS console How to fix it ? BG Dome 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 ___ 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 retain the B-number in an Call forward
Have you tried setting the effective_caller_id_number before bridging? Such as: action application=set data=effective_caller_id_number=9185551212/ Cheers, Rob On Oct 20, 2009, at 6:55 AM, Durk de Beer wrote: Hello, I us a call forward on freeswitch to forward calls to my mobile phone. If now Freeswitch forwards a call, the number information presented to my mobile phone is that of the original caller, because that is what there is appearing in its display. What I want to see is that Freeswitch is calling me so I would know it is a business call and not an private call. Now I have no means to distinguishes between them. Kind regards, Durk de Beer ___ 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_odbc_query share memory problem
Hi, I haven't run into that problem yet, but did you try increasing the maximum shared memory in /proc/sys/kernel/shmmax (sysctl kernel.shmmax) ? regards, Leon On Oct 20, 2009, at 3:31 PM, Dome Charoenyost wrote: Dear Sir, I'm using mod_odbc_query and mod_nibble_billing for my calling card solutoin. i found mod_odbc_query cant work with high load (200 calls concurrent) i got error [STATE: 53200 CODE 7 ERROR: [unixODBC]ERROR: out of shared memory; Error while executing the query ] in FS console How to fix it ? BG Dome 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 ___ 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_odbc_query share memory problem
Try 300 seconds (5 minutes) and see if it improves. On Oct 20, 2009, at 9:00 AM, Dome Charoenyost wrote: 2009/10/20 Rob Forman rob4manh...@gmail.com: Hi Dome, Have you tried increasing global_heartbeat to reduce the frequency of odbc calls? What is it currently set to? Now 1 min Rob On Oct 20, 2009, at 8:31 AM, Dome Charoenyost wrote: Dear Sir, I'm using mod_odbc_query and mod_nibble_billing for my calling card solutoin. i found mod_odbc_query cant work with high load (200 calls concurrent) i got error [STATE: 53200 CODE 7 ERROR: [unixODBC]ERROR: out of shared memory; Error while executing the query ] in FS console How to fix it ? BG Dome 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 ___ 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] can't dial from IPv6 to IPv6
No, it's the same thing with /64 on both ends. I tried to built SVN version and modified just IP addresses in sip_profiles, but still can't call from one phone to another. It goes straght to voicemail. Is there a softphone, which you know works with freeswitch on IPv6? Maybe the error is in phones I'm currently using. On Tue, Oct 20, 2009 at 3:34 PM, Dan White dwh...@olp.net wrote: This address caught my eye. Although valid, a /32 address assignment is atypical. Is that what you intended? Does assigning a /64 address on both ends make any difference? -- Dan White ___ 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] can't dial from IPv6 to IPv6
Well how are you trying to dial users? /b On Oct 20, 2009, at 9:16 AM, ineya ineya wrote: No, it's the same thing with /64 on both ends. I tried to built SVN version and modified just IP addresses in sip_profiles, but still can't call from one phone to another. It goes straght to voicemail. Is there a softphone, which you know works with freeswitch on IPv6? Maybe the error is in phones I'm currently using. ___ 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] Is anyone running Ubuntu 8.04/Hardy?
Gabriel Gunderson wrote: On Mon, Oct 19, 2009 at 11:35 AM, Mark Sobkow m.sob...@marketelsystems.com wrote: Everyone I've emailed with on the dev list is running the current release of Ubuntu, not 8.04/Hardy. Well, what issues? Gabe ___ 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 We had problems with loading mod_sofia. However, we were running an older release, then we'd tried upgrading to the source build from launchpad.net. Yesterday we downloaded the current svn.freeswitch.org, and that particular problem has gone away. Now we need to figure out why our Erlang component can synchronize from Windows, but not from Linux. -- Mark Sobkow Senior Developer MarkeTel Multi-Line Dialing Systems LTD. 428 Victoria Ave Regina, SK S4N-0P6 Toll-Free: 800-289-8616-X533 Local: 306-359-6893-X533 Fax: 306-359-6879 Email: m.sob...@marketelsystems.com Web: http://www.marketelsystems.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
Re: [Freeswitch-users] mod_odbc_query share memory problem
2009/10/20 Rob Forman rob4manh...@gmail.com: Try 300 seconds (5 minutes) and see if it improves. Ok. i'll try On Oct 20, 2009, at 9:00 AM, Dome Charoenyost wrote: 2009/10/20 Rob Forman rob4manh...@gmail.com: Hi Dome, Have you tried increasing global_heartbeat to reduce the frequency of odbc calls? What is it currently set to? Now 1 min Rob On Oct 20, 2009, at 8:31 AM, Dome Charoenyost wrote: Dear Sir, I'm using mod_odbc_query and mod_nibble_billing for my calling card solutoin. i found mod_odbc_query cant work with high load (200 calls concurrent) i got error [STATE: 53200 CODE 7 ERROR: [unixODBC]ERROR: out of shared memory; Error while executing the query ] in FS console How to fix it ? BG Dome 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 ___ 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_odbc_query share memory problem
2009/10/20 Leon de Rooij l...@scarlet-internet.nl: Hi, I haven't run into that problem yet, but did you try increasing the maximum shared memory in /proc/sys/kernel/shmmax (sysctl kernel.shmmax) ? Recommend me please i have 4GB RAM. is posible to increase to 50% RAM ? Dome C. regards, Leon On Oct 20, 2009, at 3:31 PM, Dome Charoenyost wrote: Dear Sir, I'm using mod_odbc_query and mod_nibble_billing for my calling card solutoin. i found mod_odbc_query cant work with high load (200 calls concurrent) i got error [STATE: 53200 CODE 7 ERROR: [unixODBC]ERROR: out of shared memory; Error while executing the query ] in FS console How to fix it ? BG Dome 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 ___ 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] Connect PHP SOAP Web Server with SQLite database of FS
Now I am building a PHP SOAP Web Service to access the database of FS. Anyone has idea about how to access sqlite database of FS through PHP ? I have read about socket event in FS, but I don't know whether it can response with the query of database or not. Thanks for your help. -- View this message in context: http://www.nabble.com/Connect-PHP-SOAP-Web-Server-with-SQLite-database-of-FS-tp25976554p25976554.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
Re: [Freeswitch-users] Fwd: mod_opal - call charged before H.225 connect
On 2009-10-20 10:17 +0200, Tihomir Culjaga wrote freeswitch-us...@lists.fre...:
TC TC
TC TCcall flow is SIP_user = FS = H323_endpoint is failing ..
TC coredumped
TC TChttp://pastebin.freeswitch.org/10703
TC
TC i fix some bugs now,
TC ftp://srv.icf.org.ru/pub/soft/f/freeswitch/mod_h323/mod_h323.tar.bz2 this
TC is
TC updated version, try it, if you experience no audio try enable rtp proxy in
TC you sip profile.
TC
TC
TCHi, there are several issues... lets start with top 4 :)
TC
TC
TC
TC1. I'm still stuck with no audio:
TC
TCI have this parameter in the sip profile set: param
TCname=inbound-proxy-media value=true/
TC...tried with both with slow start and fast start... any idea ?
TC
TCpls check: http://pastebin.freeswitch.org/10771
try to sniff rtp traffic on fs, is it go thouch him.
TC
TC
TC
TC
TC
TC2. outgoing calls still failing in coredumps: what is your dialplan ? ...
TChow do you call bridge application?
simple:
action application=bridge data=h323/${number}/
if fs not registered on gk then data=h323/${numb...@xxx.xxx.xxx.xxx.
TC
TC2009-10-20 10:08:18.424083 [DEBUG] h323caps.cxx:3252 Found capability:
TCUserInput/PointDevice 14
TC2009-10-20 10:08:18.424083 [DEBUG] h323caps.cxx:3248 FindCapability: 15
TC2009-10-20 10:08:18.426608 [DEBUG] osutils.cxx:60 Created external thread
TC0xb6eb60a0 for id 3048876944
TC2009-10-20 10:08:18.426608 [DEBUG] osutils.cxx:65 Destroyed external thread
TC0xb6ebafa8 for id 3048876944
TC2009-10-20 10:08:18.426608 [DEBUG] tlibthrd.cxx:406 Destroyed thread
TC0xb6ebafa8 PExternalThread:0xb5ba2b90(id = b5ba2b90)
TC2009-10-20 10:08:18.426608 [DEBUG] h323caps.cxx:3252 Found capability:
TCUserInput/Modal 15
TC2009-10-20 10:08:18.426608 [DEBUG] osutils.cxx:880 MONITOR: timers=0,
TCexpiries=0
TC2009-10-20 10:08:18.426608 [DEBUG] osutils.cxx:880 MONITOR: timers=0,
TCexpiries=0
TC2009-10-20 10:08:18.426608 [DEBUG] osutils.cxx:60 Created external thread
TC0xb6eba910 for id 3048876944
TC2009-10-20 10:08:18.426608 [DEBUG] h4601.cxx:1725 Endpoint Attached
TCSegmentation fault (core dumped)
TCtculj...@subzero:~/freeswitch-trunk$
TC
TCpls check: http://pastebin.freeswitch.org/10769
look strange, what version of libpt/h323plus you use and freeswitch itself ?
TC
TC
TC
TC3. when you hangup from SIP side, the call is not released end-to-end (the
TCH323 endpoint doesn't get any releaseComplete message)
TC
TC2009-10-20 10:10:51.264527 [DEBUG] osutils.cxx:880 MONITOR: timers=2,
TCexpiries=3
TC2009-10-20 10:10:51.264527 [DEBUG] h323neg.cxx:432 Received
TCMasterSlaveDeterminationAck: state=Incoming
TC2009-10-20 10:10:51.264527 [DEBUG] osutils.cxx:880 MONITOR: timers=1,
TCexpiries=3
TC2009-10-20 10:10:51.264527 [DEBUG] osutils.cxx:880 MONITOR: timers=2,
TCexpiries=4
TC2009-10-20 10:10:51.264527 [DEBUG] h323.cxx:4138
TCInternalEstablishedConnectionCheck: connectionState=EstablishedConnection
TCfastStartState=FastStartAcknowledged
TC2009-10-20 10:10:51.264527 [DEBUG] h323caps.cxx:3264 FindCapability: T.120
TC2009-10-20 10:10:51.264527 [DEBUG] h323.cxx:4138
TCInternalEstablishedConnectionCheck: connectionState=EstablishedConnection
TCfastStartState=FastStartAcknowledged
TC2009-10-20 10:10:51.264527 [DEBUG] tlibthrd.cxx:1023
TCPThread::PXBlockOnIO(45,0)
TC2009-10-20 10:10:54.405479 [NOTICE] sofia.c:328 Hangup sofia/internal/
TCsip:1...@10.4.62.89 sip%3a1...@10.4.62.89 [CS_CONSUME_MEDIA]
TC[NORMAL_CLEARING]
TC2009-10-20 10:10:54.405479 [DEBUG] switch_channel.c:1726 Send signal
TCsofia/internal/sip:1...@10.4.62.89 sip%3a1...@10.4.62.89 [KILL]
TC2009-10-20 10:10:54.405479 [DEBUG] switch_core_session.c:932 Send signal
TCsofia/internal/sip:1...@10.4.62.89 sip%3a1...@10.4.62.89 [BREAK]
TC2009-10-20 10:10:54.405479 [DEBUG] switch_core_state_machine.c:437 thread
TCmismatch skipping state handler.
TC2009-10-20 10:10:54.405479 [DEBUG] switch_core_state_machine.c:306
TC(sofia/internal/sip:1...@10.4.62.89 sip%3a1...@10.4.62.89) Running State
TCChange CS_HANGUP
TC2009-10-20 10:10:54.406530 [DEBUG] switch_core_state_machine.c:464
TC(sofia/internal/sip:1...@10.4.62.89 sip%3a1...@10.4.62.89) State HANGUP
TC2009-10-20 10:10:54.406530 [DEBUG] mod_sofia.c:338 Channel sofia/internal/
TCsip:1...@10.4.62.89 sip%3a1...@10.4.62.89 hanging up, cause:
TCNORMAL_CLEARING
TC2009-10-20 10:10:54.406530 [DEBUG] switch_core_state_machine.c:46
TCsofia/internal/sip:1...@10.4.62.89 sip%3a1...@10.4.62.89 Standard HANGUP,
TCcause: NORMAL_CLEARING
TC
TC
TCpls check: http://pastebin.freeswitch.org/10771
hm, according this log it send release complete:
2009-10-20 10:11:03.995707 [DEBUG] h323pdu.cxx:609 Sending PDU
[ip$10.4.62.7:1720/ip$10.1.14.153:3341]
i think please try to dump full traffic of this call from both endpoints and
send me.
TC
TC
TC
TC
TC4. sometimes when i shutdown FS i get coredimps - from my experience it
TClooks like you don't wait for a FS thread to finish when you exit...
TC
TC2009-10-20 10:05:59.493306 [CONSOLE] switch_event.c:508 Stopping queue
TCthread 2
TC2009-10-20 10:05:59.493339 [CONSOLE]
Re: [Freeswitch-users] Fwd: mod_opal - call charged before H.225 connect
On 2009-10-20 10:17 +0200, Tihomir Culjaga wrote freeswitch-us...@lists.fre...:
TC TC
TC TCcall flow is SIP_user = FS = H323_endpoint is failing ..
TC coredumped
TC TChttp://pastebin.freeswitch.org/10703
TC
TC i fix some bugs now,
TC ftp://srv.icf.org.ru/pub/soft/f/freeswitch/mod_h323/mod_h323.tar.bz2 this
TC is
TC updated version, try it, if you experience no audio try enable rtp proxy in
TC you sip profile.
TC
TC
TCHi, there are several issues... lets start with top 4 :)
TC
TC
TC
TC1. I'm still stuck with no audio:
TC
TCI have this parameter in the sip profile set: param
TCname=inbound-proxy-media value=true/
TC...tried with both with slow start and fast start... any idea ?
TC
TCpls check: http://pastebin.freeswitch.org/10771
try to sniff rtp traffic on fs, is it go thouch him.
TC
TC
TC
TC
TC
TC2. outgoing calls still failing in coredumps: what is your dialplan ? ...
TChow do you call bridge application?
simple:
action application=bridge data=h323/${number}/
if fs not registered on gk then data=h323/${numb...@xxx.xxx.xxx.xxx.
TC
TC2009-10-20 10:08:18.424083 [DEBUG] h323caps.cxx:3252 Found capability:
TCUserInput/PointDevice 14
TC2009-10-20 10:08:18.424083 [DEBUG] h323caps.cxx:3248 FindCapability: 15
TC2009-10-20 10:08:18.426608 [DEBUG] osutils.cxx:60 Created external thread
TC0xb6eb60a0 for id 3048876944
TC2009-10-20 10:08:18.426608 [DEBUG] osutils.cxx:65 Destroyed external thread
TC0xb6ebafa8 for id 3048876944
TC2009-10-20 10:08:18.426608 [DEBUG] tlibthrd.cxx:406 Destroyed thread
TC0xb6ebafa8 PExternalThread:0xb5ba2b90(id = b5ba2b90)
TC2009-10-20 10:08:18.426608 [DEBUG] h323caps.cxx:3252 Found capability:
TCUserInput/Modal 15
TC2009-10-20 10:08:18.426608 [DEBUG] osutils.cxx:880 MONITOR: timers=0,
TCexpiries=0
TC2009-10-20 10:08:18.426608 [DEBUG] osutils.cxx:880 MONITOR: timers=0,
TCexpiries=0
TC2009-10-20 10:08:18.426608 [DEBUG] osutils.cxx:60 Created external thread
TC0xb6eba910 for id 3048876944
TC2009-10-20 10:08:18.426608 [DEBUG] h4601.cxx:1725 Endpoint Attached
TCSegmentation fault (core dumped)
TCtculj...@subzero:~/freeswitch-trunk$
TC
TCpls check: http://pastebin.freeswitch.org/10769
look strange, what version of libpt/h323plus you use and freeswitch itself ?
TC
TC
TC
TC3. when you hangup from SIP side, the call is not released end-to-end (the
TCH323 endpoint doesn't get any releaseComplete message)
TC
TC2009-10-20 10:10:51.264527 [DEBUG] osutils.cxx:880 MONITOR: timers=2,
TCexpiries=3
TC2009-10-20 10:10:51.264527 [DEBUG] h323neg.cxx:432 Received
TCMasterSlaveDeterminationAck: state=Incoming
TC2009-10-20 10:10:51.264527 [DEBUG] osutils.cxx:880 MONITOR: timers=1,
TCexpiries=3
TC2009-10-20 10:10:51.264527 [DEBUG] osutils.cxx:880 MONITOR: timers=2,
TCexpiries=4
TC2009-10-20 10:10:51.264527 [DEBUG] h323.cxx:4138
TCInternalEstablishedConnectionCheck: connectionState=EstablishedConnection
TCfastStartState=FastStartAcknowledged
TC2009-10-20 10:10:51.264527 [DEBUG] h323caps.cxx:3264 FindCapability: T.120
TC2009-10-20 10:10:51.264527 [DEBUG] h323.cxx:4138
TCInternalEstablishedConnectionCheck: connectionState=EstablishedConnection
TCfastStartState=FastStartAcknowledged
TC2009-10-20 10:10:51.264527 [DEBUG] tlibthrd.cxx:1023
TCPThread::PXBlockOnIO(45,0)
TC2009-10-20 10:10:54.405479 [NOTICE] sofia.c:328 Hangup sofia/internal/
TCsip:1...@10.4.62.89 sip%3a1...@10.4.62.89 [CS_CONSUME_MEDIA]
TC[NORMAL_CLEARING]
TC2009-10-20 10:10:54.405479 [DEBUG] switch_channel.c:1726 Send signal
TCsofia/internal/sip:1...@10.4.62.89 sip%3a1...@10.4.62.89 [KILL]
TC2009-10-20 10:10:54.405479 [DEBUG] switch_core_session.c:932 Send signal
TCsofia/internal/sip:1...@10.4.62.89 sip%3a1...@10.4.62.89 [BREAK]
TC2009-10-20 10:10:54.405479 [DEBUG] switch_core_state_machine.c:437 thread
TCmismatch skipping state handler.
TC2009-10-20 10:10:54.405479 [DEBUG] switch_core_state_machine.c:306
TC(sofia/internal/sip:1...@10.4.62.89 sip%3a1...@10.4.62.89) Running State
TCChange CS_HANGUP
TC2009-10-20 10:10:54.406530 [DEBUG] switch_core_state_machine.c:464
TC(sofia/internal/sip:1...@10.4.62.89 sip%3a1...@10.4.62.89) State HANGUP
TC2009-10-20 10:10:54.406530 [DEBUG] mod_sofia.c:338 Channel sofia/internal/
TCsip:1...@10.4.62.89 sip%3a1...@10.4.62.89 hanging up, cause:
TCNORMAL_CLEARING
TC2009-10-20 10:10:54.406530 [DEBUG] switch_core_state_machine.c:46
TCsofia/internal/sip:1...@10.4.62.89 sip%3a1...@10.4.62.89 Standard HANGUP,
TCcause: NORMAL_CLEARING
TC
TC
TCpls check: http://pastebin.freeswitch.org/10771
hm, according this log it send release complete:
2009-10-20 10:11:03.995707 [DEBUG] h323pdu.cxx:609 Sending PDU
[ip$10.4.62.7:1720/ip$10.1.14.153:3341]
i think please try to dump full traffic of this call from both endpoints and
send me.
TC
TC
TC
TC
TC4. sometimes when i shutdown FS i get coredimps - from my experience it
TClooks like you don't wait for a FS thread to finish when you exit...
TC
TC2009-10-20 10:05:59.493306 [CONSOLE] switch_event.c:508 Stopping queue
TCthread 2
TC2009-10-20 10:05:59.493339 [CONSOLE]
Re: [Freeswitch-users] mod_odbc_query share memory problem
You could, but I would try just doubling whatever it is to see if thats improves the issue first. The default is 32MB. You could double it to 64MB and test again. What is it currently set to (run: sysctl kernel.shmmax)? You can change it on the fly with sysctl. Once you're done testing and want it to persist across reboots, you'll want to set in /etc/sysctl.conf . Cheers, Rob On Oct 20, 2009, at 9:35 AM, Dome Charoenyost wrote: 2009/10/20 Leon de Rooij l...@scarlet-internet.nl: Hi, I haven't run into that problem yet, but did you try increasing the maximum shared memory in /proc/sys/kernel/shmmax (sysctl kernel.shmmax) ? Recommend me please i have 4GB RAM. is posible to increase to 50% RAM ? Dome C. regards, Leon On Oct 20, 2009, at 3:31 PM, Dome Charoenyost wrote: Dear Sir, I'm using mod_odbc_query and mod_nibble_billing for my calling card solutoin. i found mod_odbc_query cant work with high load (200 calls concurrent) i got error [STATE: 53200 CODE 7 ERROR: [unixODBC]ERROR: out of shared memory; Error while executing the query ] in FS console How to fix it ? BG Dome 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 ___ 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_odbc_query share memory problem
2009/10/20 Rob Forman rob4manh...@gmail.com: You could, but I would try just doubling whatever it is to see if thats improves the issue first. The default is 32MB. You could double it to 64MB and test again. Ok' let's me try 64MB first i found some information https://www.ishoof.org/project/wiki/index.php/ORACLE_11_G_-_Gentoo_Installation What is it currently set to (run: sysctl kernel.shmmax)? You can change it on the fly with sysctl. Once you're done testing and want it to persist across reboots, you'll want to set in /etc/sysctl.conf . Cheers, Rob On Oct 20, 2009, at 9:35 AM, Dome Charoenyost wrote: 2009/10/20 Leon de Rooij l...@scarlet-internet.nl: Hi, I haven't run into that problem yet, but did you try increasing the maximum shared memory in /proc/sys/kernel/shmmax (sysctl kernel.shmmax) ? Recommend me please i have 4GB RAM. is posible to increase to 50% RAM ? Dome C. regards, Leon On Oct 20, 2009, at 3:31 PM, Dome Charoenyost wrote: Dear Sir, I'm using mod_odbc_query and mod_nibble_billing for my calling card solutoin. i found mod_odbc_query cant work with high load (200 calls concurrent) i got error [STATE: 53200 CODE 7 ERROR: [unixODBC]ERROR: out of shared memory; Error while executing the query ] in FS console How to fix it ? BG Dome 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 ___ 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] Fwd: mod_opal - call charged before H.225 connect
On 2009-10-20 10:17 +0200, Tihomir Culjaga wrote freeswitch-us...@lists.fre...:
TC TC
TC TCcall flow is SIP_user = FS = H323_endpoint is failing ..
TC coredumped
TC TChttp://pastebin.freeswitch.org/10703
TC
TC i fix some bugs now,
TC ftp://srv.icf.org.ru/pub/soft/f/freeswitch/mod_h323/mod_h323.tar.bz2 this
TC is
TC updated version, try it, if you experience no audio try enable rtp proxy in
TC you sip profile.
TC
TC
TCHi, there are several issues... lets start with top 4 :)
TC
TC
TC
TC1. I'm still stuck with no audio:
TC
TCI have this parameter in the sip profile set: param
TCname=inbound-proxy-media value=true/
TC...tried with both with slow start and fast start... any idea ?
TC
TCpls check: http://pastebin.freeswitch.org/10771
try to sniff rtp traffic on fs, is it go thouch him.
TC
TC
TC
TC
TC
TC2. outgoing calls still failing in coredumps: what is your dialplan ? ...
TChow do you call bridge application?
simple:
action application=bridge data=h323/${number}/
if fs not registered on gk then data=h323/${numb...@xxx.xxx.xxx.xxx.
TC
TC2009-10-20 10:08:18.424083 [DEBUG] h323caps.cxx:3252 Found capability:
TCUserInput/PointDevice 14
TC2009-10-20 10:08:18.424083 [DEBUG] h323caps.cxx:3248 FindCapability: 15
TC2009-10-20 10:08:18.426608 [DEBUG] osutils.cxx:60 Created external thread
TC0xb6eb60a0 for id 3048876944
TC2009-10-20 10:08:18.426608 [DEBUG] osutils.cxx:65 Destroyed external thread
TC0xb6ebafa8 for id 3048876944
TC2009-10-20 10:08:18.426608 [DEBUG] tlibthrd.cxx:406 Destroyed thread
TC0xb6ebafa8 PExternalThread:0xb5ba2b90(id = b5ba2b90)
TC2009-10-20 10:08:18.426608 [DEBUG] h323caps.cxx:3252 Found capability:
TCUserInput/Modal 15
TC2009-10-20 10:08:18.426608 [DEBUG] osutils.cxx:880 MONITOR: timers=0,
TCexpiries=0
TC2009-10-20 10:08:18.426608 [DEBUG] osutils.cxx:880 MONITOR: timers=0,
TCexpiries=0
TC2009-10-20 10:08:18.426608 [DEBUG] osutils.cxx:60 Created external thread
TC0xb6eba910 for id 3048876944
TC2009-10-20 10:08:18.426608 [DEBUG] h4601.cxx:1725 Endpoint Attached
TCSegmentation fault (core dumped)
TCtculj...@subzero:~/freeswitch-trunk$
TC
TCpls check: http://pastebin.freeswitch.org/10769
look strange, what version of libpt/h323plus you use and freeswitch itself ?
TC
TC
TC
TC3. when you hangup from SIP side, the call is not released end-to-end (the
TCH323 endpoint doesn't get any releaseComplete message)
TC
TC2009-10-20 10:10:51.264527 [DEBUG] osutils.cxx:880 MONITOR: timers=2,
TCexpiries=3
TC2009-10-20 10:10:51.264527 [DEBUG] h323neg.cxx:432 Received
TCMasterSlaveDeterminationAck: state=Incoming
TC2009-10-20 10:10:51.264527 [DEBUG] osutils.cxx:880 MONITOR: timers=1,
TCexpiries=3
TC2009-10-20 10:10:51.264527 [DEBUG] osutils.cxx:880 MONITOR: timers=2,
TCexpiries=4
TC2009-10-20 10:10:51.264527 [DEBUG] h323.cxx:4138
TCInternalEstablishedConnectionCheck: connectionState=EstablishedConnection
TCfastStartState=FastStartAcknowledged
TC2009-10-20 10:10:51.264527 [DEBUG] h323caps.cxx:3264 FindCapability: T.120
TC2009-10-20 10:10:51.264527 [DEBUG] h323.cxx:4138
TCInternalEstablishedConnectionCheck: connectionState=EstablishedConnection
TCfastStartState=FastStartAcknowledged
TC2009-10-20 10:10:51.264527 [DEBUG] tlibthrd.cxx:1023
TCPThread::PXBlockOnIO(45,0)
TC2009-10-20 10:10:54.405479 [NOTICE] sofia.c:328 Hangup sofia/internal/
TCsip:1...@10.4.62.89 sip%3a1...@10.4.62.89 [CS_CONSUME_MEDIA]
TC[NORMAL_CLEARING]
TC2009-10-20 10:10:54.405479 [DEBUG] switch_channel.c:1726 Send signal
TCsofia/internal/sip:1...@10.4.62.89 sip%3a1...@10.4.62.89 [KILL]
TC2009-10-20 10:10:54.405479 [DEBUG] switch_core_session.c:932 Send signal
TCsofia/internal/sip:1...@10.4.62.89 sip%3a1...@10.4.62.89 [BREAK]
TC2009-10-20 10:10:54.405479 [DEBUG] switch_core_state_machine.c:437 thread
TCmismatch skipping state handler.
TC2009-10-20 10:10:54.405479 [DEBUG] switch_core_state_machine.c:306
TC(sofia/internal/sip:1...@10.4.62.89 sip%3a1...@10.4.62.89) Running State
TCChange CS_HANGUP
TC2009-10-20 10:10:54.406530 [DEBUG] switch_core_state_machine.c:464
TC(sofia/internal/sip:1...@10.4.62.89 sip%3a1...@10.4.62.89) State HANGUP
TC2009-10-20 10:10:54.406530 [DEBUG] mod_sofia.c:338 Channel sofia/internal/
TCsip:1...@10.4.62.89 sip%3a1...@10.4.62.89 hanging up, cause:
TCNORMAL_CLEARING
TC2009-10-20 10:10:54.406530 [DEBUG] switch_core_state_machine.c:46
TCsofia/internal/sip:1...@10.4.62.89 sip%3a1...@10.4.62.89 Standard HANGUP,
TCcause: NORMAL_CLEARING
TC
TC
TCpls check: http://pastebin.freeswitch.org/10771
hm, according this log it send release complete:
2009-10-20 10:11:03.995707 [DEBUG] h323pdu.cxx:609 Sending PDU
[ip$10.4.62.7:1720/ip$10.1.14.153:3341]
i think please try to dump full traffic of this call from both endpoints and
send me.
TC
TC
TC
TC
TC4. sometimes when i shutdown FS i get coredimps - from my experience it
TClooks like you don't wait for a FS thread to finish when you exit...
TC
TC2009-10-20 10:05:59.493306 [CONSOLE] switch_event.c:508 Stopping queue
TCthread 2
TC2009-10-20 10:05:59.493339 [CONSOLE]
Re: [Freeswitch-users] can't dial from IPv6 to IPv6
By numbers, I have 2 numbers registered, so I dial 1006 or 1007. This worked for IPv4, so I haven't thought about doing it differently for IPv6. SIP messages looked OK to me - you just gave me an idea to compare SIP messages for IPv4 and IPv6. I'll try that tommorow. On Tue, Oct 20, 2009 at 4:21 PM, Brian West br...@freeswitch.org wrote: Well how are you trying to dial users? /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] mod_odbc_query share memory problem
Those are recommended values for Oracle. Freeswitch != Oracle. They behave and use a server's resources very differently. I wouldn't change more than you need for now. Tweak shmmax then go from there. Cheers, Rob On Oct 20, 2009, at 10:14 AM, Dome Charoenyost wrote: 2009/10/20 Rob Forman rob4manh...@gmail.com: You could, but I would try just doubling whatever it is to see if thats improves the issue first. The default is 32MB. You could double it to 64MB and test again. Ok' let's me try 64MB first i found some information https://www.ishoof.org/project/wiki/index.php/ORACLE_11_G_-_Gentoo_Installation What is it currently set to (run: sysctl kernel.shmmax)? You can change it on the fly with sysctl. Once you're done testing and want it to persist across reboots, you'll want to set in /etc/ sysctl.conf . Cheers, Rob On Oct 20, 2009, at 9:35 AM, Dome Charoenyost wrote: 2009/10/20 Leon de Rooij l...@scarlet-internet.nl: Hi, I haven't run into that problem yet, but did you try increasing the maximum shared memory in /proc/sys/kernel/shmmax (sysctl kernel.shmmax) ? Recommend me please i have 4GB RAM. is posible to increase to 50% RAM ? Dome C. regards, Leon On Oct 20, 2009, at 3:31 PM, Dome Charoenyost wrote: Dear Sir, I'm using mod_odbc_query and mod_nibble_billing for my calling card solutoin. i found mod_odbc_query cant work with high load (200 calls concurrent) i got error [STATE: 53200 CODE 7 ERROR: [unixODBC]ERROR: out of shared memory; Error while executing the query ] in FS console How to fix it ? BG Dome 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 ___ 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] Is anyone running Ubuntu 8.04/Hardy?
maybe next time test and/or search the mailing list before asking. I was a little worried when I read that it do not works on Hardy. Good to be reassured, it works. :-) On 10/20/09, Mark Sobkow m.sob...@marketelsystems.com wrote: Gabriel Gunderson wrote: On Mon, Oct 19, 2009 at 11:35 AM, Mark Sobkow m.sob...@marketelsystems.com wrote: Everyone I've emailed with on the dev list is running the current release of Ubuntu, not 8.04/Hardy. Well, what issues? Gabe ___ 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 We had problems with loading mod_sofia. However, we were running an older release, then we'd tried upgrading to the source build from launchpad.net. Yesterday we downloaded the current svn.freeswitch.org, and that particular problem has gone away. Now we need to figure out why our Erlang component can synchronize from Windows, but not from Linux. -- Mark Sobkow Senior Developer MarkeTel Multi-Line Dialing Systems LTD. 428 Victoria Ave Regina, SK S4N-0P6 Toll-Free: 800-289-8616-X533 Local: 306-359-6893-X533 Fax: 306-359-6879 Email: m.sob...@marketelsystems.com Web: http://www.marketelsystems.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 -- Sent from my mobile device 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
Re: [Freeswitch-users] bind_meta_app problem with custome extension
OMG such a stupid user error!
I have to write
action application=bind_meta_app data=7 ab s
/usr/local/freeswitch/conf/dialplan/execute_extension::roar XML features/
with double colon instead of a single one.
Thanks everyone for help!
Uncle Johny wrote:
Hi guys,
I hope you can help me with this one since it looks like some FS bug or my
user error :) .I have a simple dialplan in FS which allows to transfer a
person to some 3rd party phone but it works only if you call it from
default context.
If it is not called from default context I get this error all the time:
2009-09-12 11:58:17.484680 [ERR] mod_native_file.c:68 Error opening
/usr/local/freeswitch/conf/dialplan/execute_extension:roar XML
features.PCMU
With the following configuration 2 parties will be connected but whenever
you press *7 it just does not work and FS splits the message. Very weird
that this works if I put the extension into default.xml.
Any ideas what is wrong here?
My config:
I tried to create my own context and call it pstn.
So when FS gets a call from proxy it goes to pstn context, since I have
this line in sip_proxies/internal.xml
in dialplan directory I have pstn.xml
include
context name=pstn
extension name=pstn_enum
condition field=destination_number expression=^(.*)$
action application=answer/
action application=bind_meta_app data=7 ab s
/usr/local/freeswitch/conf/dialplan/execute_extension:roar XML features/
action application=set
data=hangup_after_bridge=true/
action application=bridge
data=sofia/external/${destination_numb...@207.88.122.16/
/condition
/extension
/context
/include
In features.xml (It can be anything there since FS never reaches this code
and dies before executing it):
extension name=roar
condition field=destination_number expression=^roar$
action application=set data=continue_on_fail=true/
action application=log data=INFO transfer context /
action application=read data=1 1 /root/1.gsm attxfer_callthis
1000 #/
action application=log data=INFO blah ${attxfer_callthis}/
action application=set data=origination_cancel_key=#/
action application=log data=INFO after read/
/condition
/extension
--
View this message in context:
http://www.nabble.com/bind_meta_app-problem-with-custome-extension-tp25963971p25977224.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.
___
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Re: [Freeswitch-users] Stereo Support
There is not enough spec/devices using spec on stereo to try to implement it at this time. Maybe some day. On Tue, Oct 20, 2009 at 2:35 AM, Helmut Kuper helmut.ku...@ewetel.dewrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hello, I wonder if there is real stereo support planned for FreeSWITCH in terms of streaming music/video-audio to desktop? I ask, because I heared that celt codec is supporting stereo. regards helmut -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (MingW32) iD8DBQFK3Whc4tZeNddg3dwRAkdEAKCLBSV4H8q5JN4NZ39fgSZPxr5PCgCgmpTq PAu8qozLP5FaVKIwfCXVqjU= =Zseo -END PGP SIGNATURE- ___ 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
Re: [Freeswitch-users] mod_odbc_query share memory problem
Now i'm setting to 64MB I found channel problem also. many channel not disconnect freeswi...@internal show calls count 87 total. freeswi...@internal show channels count 466 total. freeswi...@internal status UP 0 years, 0 days, 1 hour, 53 minutes, 23 seconds, 872 milliseconds, 0 microseconds 11641 session(s) since startup 154 session(s) 0/30 1000 session(s) max I think channels should be 154 (same session) but 466 is to high. may be some channels not disconnect Dome C. 2009/10/20 Rob Forman rob4manh...@gmail.com: Those are recommended values for Oracle. Freeswitch != Oracle. They behave and use a server's resources very differently. I wouldn't change more than you need for now. Tweak shmmax then go from there. Cheers, Rob On Oct 20, 2009, at 10:14 AM, Dome Charoenyost wrote: 2009/10/20 Rob Forman rob4manh...@gmail.com: You could, but I would try just doubling whatever it is to see if thats improves the issue first. The default is 32MB. You could double it to 64MB and test again. Ok' let's me try 64MB first i found some information https://www.ishoof.org/project/wiki/index.php/ORACLE_11_G_-_Gentoo_Installation What is it currently set to (run: sysctl kernel.shmmax)? You can change it on the fly with sysctl. Once you're done testing and want it to persist across reboots, you'll want to set in /etc/ sysctl.conf . Cheers, Rob On Oct 20, 2009, at 9:35 AM, Dome Charoenyost wrote: 2009/10/20 Leon de Rooij l...@scarlet-internet.nl: Hi, I haven't run into that problem yet, but did you try increasing the maximum shared memory in /proc/sys/kernel/shmmax (sysctl kernel.shmmax) ? Recommend me please i have 4GB RAM. is posible to increase to 50% RAM ? Dome C. regards, Leon On Oct 20, 2009, at 3:31 PM, Dome Charoenyost wrote: Dear Sir, I'm using mod_odbc_query and mod_nibble_billing for my calling card solutoin. i found mod_odbc_query cant work with high load (200 calls concurrent) i got error [STATE: 53200 CODE 7 ERROR: [unixODBC]ERROR: out of shared memory; Error while executing the query ] in FS console How to fix it ? BG Dome 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 ___ 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 ___ 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] can't dial from IPv6 to IPv6
The defaults will NOT work with ipv6 out of the box because the sofia_contact on the directory only looks at the internal profile NOT the internal-ipv6 profile... open up the directory default and change the sofia_contact to prepend the internal-ipv6/u...@domain /b On Oct 20, 2009, at 10:21 AM, ineya ineya wrote: By numbers, I have 2 numbers registered, so I dial 1006 or 1007. This worked for IPv4, so I haven't thought about doing it differently for IPv6. SIP messages looked OK to me - you just gave me an idea to compare SIP messages for IPv4 and IPv6. I'll try that tommorow. ___ 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] CS_REPORTING Channel event state
Dear All What's CS_REPORTING state ? I found many channels not hang up ans state is CS_REPORTING e264f84a-bd87-11de-9a90-2320c02172de,outbound,2009-10-20 21:50:26,1256050226,sofia/external/x...@xxx.xxx.xx.191:7050,CS_REPORTING,FreeSWITCH,,xx.xxx.xxx.xxx,7050,,,XML,public, BG Dome 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] Qustion about INFO messages after Connect/Answer
On Tue, Oct 20, 2009 at 12:25 AM, Helmut Kuper helmut.ku...@ewetel.dewrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hello Anthony,
On 19.10.2009 22:07, Anthony Minessale wrote:
please update and test trunk
1) I changed the core to remove the excess data by default in your
scenario
2) I added variables you can use to control it
origination_callee_id_name origination_callee_id_number which belong in
{} in the dial string eg
{origination_callee_id_number=1234}openzap/1/a/1234
updated and testet for SIP calls. origination_callee_id_number=1234
works within the dial string and via using export application but not
via set application. Didn't test it for openzap, yet but I guess it
will work, too.
Thanks for testing. BTW, according to Tony's instructions the user needs to
put the variable definition inside the {} of the dialstring. This means that
the vars are designed for the new leg, not the local leg, which means that
export would work but set would not. (Remember, set is to set a chan
var on the local channel, export will set a chan var locally *and* on the
other call leg. See also the nolocal option of export.)
What left is that unknown thing on callee's display (SIP phones only I
guess) ... I would suggest to make sending unknown INFO message
optional, but to be honest, I have no idea for what it was invented, so
maybe my suggestion is nonsens.
Under what conditions did you see unknown? I'm wondering if the user can
just pick a default other than unknown if he wants something else to be
displayed.
Thoughts?
-MC
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Re: [Freeswitch-users] Qustion about INFO messages after Connect/Answer
Or just set the var to what you want it to say? /b On Oct 20, 2009, at 11:19 AM, Michael Collins wrote: Under what conditions did you see unknown? I'm wondering if the user can just pick a default other than unknown if he wants something else to be displayed. Thoughts? -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
Re: [Freeswitch-users] can't dial from IPv6 to IPv6
Hmm, I didn't about this.
so in directory/default.xml I have:
param name=dial-string
value={presence_id=${dialed_us...@${dialed_domain}}${sofia_contact(${dialed_us...@${dialed_domain})}/
and the modified version for IPv6 would be ?...
param name=dial-string
value={presence_id=${dialed_us...@${dialed_domain}}${sofia_contact(internal-ipv6/${dialed_us...@${dialed_domain})}/
On Tue, Oct 20, 2009 at 5:34 PM, Brian West br...@freeswitch.org wrote:
The defaults will NOT work with ipv6 out of the box because the
sofia_contact on the directory only looks at the internal profile NOT
the internal-ipv6 profile... open up the directory default and change
the sofia_contact to prepend the internal-ipv6/u...@domain
/b
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[Freeswitch-users] FreeSWITCH Weekly Conference Call Agenda - Oct 23rd
FYI, The conference call agenda for this week is online: http://wiki.freeswitch.org/wiki/FS_weekly_2009_10_23 Also, we will be discussing whether or not Friday is the best time to be doing this call. Just remember that the FS devs have lots of work to do and they can't always accommodate everyone's schedules. Still, they'll do their best to be flexible wherever possible. Thanks for supporting the weekly conference call. -Michael ___ 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] Janitorial Project Update - Wiki Cleanup
Hello all, As you may know we have a weekly conference call each Friday. This week's agenda has several wiki cleanup sub-projects. I wanted to crowdsource this because, as the saying goes, many hands makes the work light. In other words, if everyone can help a little bit then we won't be dumping a lot of work on just a few people. HOW YOU CAN HELP If you don't already have a wiki account then please go to http://wiki.freeswitch.org and create one. All wiki users have editing rights and can make changes to wiki content. Don't worry about making mistakes! We can always revert a change. Brian West and I review all wiki edits each day. Standard users can't accidentally delete a page or permanently remove content. You're totally safe to make edits. By the way, when you sign in to the wiki you won't see the ads on the left-hand side of the page and you won't have to scroll down so far to locate the search box. There are several ways that you can assist with the wiki cleanup. One way is to help users like Diego Viola who are quite literally going page to page looking for spelling errors, grammar, etc. If you see an obvious error please by all means correct it. Another thing you can do is visit this page and look at the janitorial section: http://wiki.freeswitch.org/wiki/FS_weekly_2009_10_23 There are several areas where we can clean up and improve the wiki: *Dead end pages - these are pages that are linked to by other wiki pages but do not themselves have links back to anywhere. At the very least each page should have a link back to some other see also kind of content. *Double redirects - these are redirect pages that point to other redirect pages. These need to be pointed to the ultimate destination page. *Orphaned pages - orphans are pages that are not linked to by other pages. The only way to access these pages is via a search or if you already know the URL. We need people to review the orphaned pages and add links to them from other pages. *Wanted pages and categories - these are pages and categories that have links but no content. We need people review them and add content or redirects as appropriate. *Uncategorized pages - all pages belong to some category and these pages haven't been added to a category yet. Categories are handy because they act as natural indexes. Please review pages that have no categories. If you have an idea for a category that does not yet exist please email me and let me know. We don't want to have too many categories. The best way to learn wiki markup is to click the edit button on a page and see what wiki markup does. You can change the markup and then click Preview to see what it doesn. Click the cancel link to discard any changes. Play around and learn! Thank you for your help. Please feel free to ask for assistance with editing the wiki. It is a community resource and the more people who know how to add content, the better off we all are. Thanks, Michael ___ 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] Status of ubuntu/debian packages.
William, Where is mod_skypiax.so? Isn't freeswitch-skypiax package expected to contain it? - Dmitry Bely ___ 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] Connect PHP SOAP Web Server with SQLite database of FS
If you really wanted: http://php.net/manual/en/book.sqlite.php But I would recommend you make use of ODBC to use a client/server RDBMS. Here's some good reading: http://www.sqlite.org/cvstrac/wiki?p=WhenToUseSqlite On October 20, 2009 10:53:01 am homqua wrote: Now I am building a PHP SOAP Web Service to access the database of FS. Anyone has idea about how to access sqlite database of FS through PHP ? I have read about socket event in FS, but I don't know whether it can response with the query of database or not. Thanks for your help. ___ 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] Digium TE220 wiki
Thanks. I added minor changes, plus a category and see also. Thanks for adding this content! -MC On Tue, Oct 20, 2009 at 12:18 AM, lakshmanan ganapathy lakindi...@gmail.com wrote: Hi all, I've created a wiki page, which contains the example configuration for making Digium TE220 to work. I request you people to check this, and give feedbacks. http://wiki.freeswitch.org/wiki/Configuration_OpenZap-DigiumTE220-Example ___ 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] Troubles with proxy media mode
Hello everyone, I'm trying to use proxy media across two profiles. The codec settings are identical, they both have late negotiation enabled, and they both have inbound-proxy-media set to true (I also tried setting proxy_media from the dialplan). FreeSWITCH ends up clearing the call with TRANSCODING_NECESSARY but I can't figure out why it thinks it needs to transcode for this call. I've attached a level 7 debug and an ngrep siptrace showing traffic from both profiles. FreeSWITCH trunk rev. 15180 running on Debian 5.0.2. See anything interesting? Thanks! -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com level7.log Description: Binary data siptrace.log Description: Binary data ___ 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] Troubles with proxy media mode
Fixed in 15181 =D 1 revision higher doh =D On Tue, Oct 20, 2009 at 2:56 PM, Kristian Kielhofner kristian.kielhof...@gmail.com wrote: Hello everyone, I'm trying to use proxy media across two profiles. The codec settings are identical, they both have late negotiation enabled, and they both have inbound-proxy-media set to true (I also tried setting proxy_media from the dialplan). FreeSWITCH ends up clearing the call with TRANSCODING_NECESSARY but I can't figure out why it thinks it needs to transcode for this call. I've attached a level 7 debug and an ngrep siptrace showing traffic from both profiles. FreeSWITCH trunk rev. 15180 running on Debian 5.0.2. See anything interesting? Thanks! -- 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 -- 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
Re: [Freeswitch-users] Troubles with proxy media mode
you are making FS to play wav file when sending a call in G711 or GSM or some other codec. you might use mod_native_filehttp://wiki.freeswitch.org/wiki/Mod_native_fileto avoid transcoding. T. On Tue, Oct 20, 2009 at 9:56 PM, Kristian Kielhofner kristian.kielhof...@gmail.com wrote: Hello everyone, I'm trying to use proxy media across two profiles. The codec settings are identical, they both have late negotiation enabled, and they both have inbound-proxy-media set to true (I also tried setting proxy_media from the dialplan). FreeSWITCH ends up clearing the call with TRANSCODING_NECESSARY but I can't figure out why it thinks it needs to transcode for this call. I've attached a level 7 debug and an ngrep siptrace showing traffic from both profiles. FreeSWITCH trunk rev. 15180 running on Debian 5.0.2. See anything interesting? Thanks! -- 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
Re: [Freeswitch-users] Troubles with proxy media mode
Updated and now working. Thanks! On Tue, Oct 20, 2009 at 4:11 PM, Anthony Minessale anthony.miness...@gmail.com wrote: Fixed in 15181 =D 1 revision higher doh =D -- 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] Proxy media mode with T.38 re-invite
Hello everyone, Now that proxy media mode is working again I'm trying to figure out why T.38 with re-INVITE doesn't... Everything goes well until my end tries to re-INVITE to T.38: U 10.16.5.129:5060 - 65.196.170.191:5060 INVITE sip:mod_so...@65.196.170.191:5060 SIP/2.0. Via: SIP/2.0/UDP 10.16.5.129:5060;branch=z9hG4bK-2656bddb. From: sip:fax_9...@10.16.5.129:5060;tag=9f75a042a3ca1f17i0. To: WIRELESS CALLER sip:19412848...@65.196.170.191;tag=4eUH874j0eUZF. Remote-Party-ID: fax_9186 sip:fax_9...@65.196.170.191;screen=yes;party=called. Call-ID: 2889ab05-385d-122d-07b2-0014221c0392. CSeq: 101 INVITE. Max-Forwards: 70. Contact: fax_9186 sip:fax_9...@10.16.5.129:5060. Expires: 30. User-Agent: Linksys/SPA3102-5.1.10(GW). Content-Length: 261. Content-Type: application/sdp. . v=0. o=- 57670 57670 IN IP4 10.16.5.129. s=-. c=IN IP4 10.16.5.129. t=0 0. m=image 16454 udptl t38. a=T38FaxVersion:0. a=T38MaxBitRate:14400. a=T38FaxRateManagement:transferredTCF. a=T38FaxMaxBuffer:200. a=T38FaxMaxDatagram:200. a=T38FaxUdpEC:t38UDPRedundancy. U 65.196.170.191:5060 - 10.16.5.129:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 10.16.5.129:5060;branch=z9hG4bK-2656bddb. From: sip:fax_9...@10.16.5.129:5060;tag=9f75a042a3ca1f17i0. To: WIRELESS CALLER sip:19412848...@65.196.170.191;tag=4eUH874j0eUZF. Call-ID: 2889ab05-385d-122d-07b2-0014221c0392. CSeq: 101 INVITE. User-Agent: Star2Star Media. Content-Length: 0. . U 65.196.170.191:5060 - 10.16.5.129:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 10.16.5.129:5060;branch=z9hG4bK-2656bddb. From: sip:fax_9...@10.16.5.129:5060;tag=9f75a042a3ca1f17i0. To: WIRELESS CALLER sip:19412848...@65.196.170.191;tag=4eUH874j0eUZF. Call-ID: 2889ab05-385d-122d-07b2-0014221c0392. CSeq: 101 INVITE. Contact: sip:mod_so...@65.196.170.191:5060. User-Agent: Star2Star Media. Accept: application/sdp. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER, REFER, UPDATE, NOTIFY, PUBLISH, SUBSCRIBE. Supported: timer, precondition, path, replaces. Content-Type: application/sdp. Content-Disposition: session. Content-Length: 242. . v=0. o=Sonus_UAC 881416742673069109 186840485790971511 IN IP4 65.196.170.129. s=SIP Media Capabilities. c=IN IP4 65.196.170.191. t=0 0. m=audio 19646 RTP/AVP 101. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=silenceSupp:off - - - -. U 10.16.5.129:5060 - 65.196.170.191:5060 ACK sip:mod_so...@65.196.170.191:5060 SIP/2.0. Via: SIP/2.0/UDP 10.16.5.129:5060;branch=z9hG4bK-842177a7. From: sip:fax_9...@10.16.5.129:5060;tag=9f75a042a3ca1f17i0. To: WIRELESS CALLER sip:19412848...@65.196.170.191;tag=4eUH874j0eUZF. Call-ID: 2889ab05-385d-122d-07b2-0014221c0392. CSeq: 101 ACK. Max-Forwards: 70. Contact: fax_9186 sip:fax_9...@10.16.5.129:5060. User-Agent: Linksys/SPA3102-5.1.10(GW). Content-Length: 0. . U 10.16.5.129:5060 - 65.196.170.191:5060 BYE sip:mod_so...@65.196.170.191:5060 SIP/2.0. Via: SIP/2.0/UDP 10.16.5.129:5060;branch=z9hG4bK-9e6b284b. From: sip:fax_9...@10.16.5.129:5060;tag=9f75a042a3ca1f17i0. To: WIRELESS CALLER sip:19412848...@65.196.170.191;tag=4eUH874j0eUZF. Call-ID: 2889ab05-385d-122d-07b2-0014221c0392. CSeq: 102 BYE. Max-Forwards: 70. User-Agent: Linksys/SPA3102-5.1.10(GW). Content-Length: 0. . U 65.196.170.191:5060 - 10.16.5.129:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 10.16.5.129:5060;branch=z9hG4bK-9e6b284b. From: sip:fax_9...@10.16.5.129:5060;tag=9f75a042a3ca1f17i0. To: WIRELESS CALLER sip:19412848...@65.196.170.191;tag=4eUH874j0eUZF. Call-ID: 2889ab05-385d-122d-07b2-0014221c0392. CSeq: 102 BYE. User-Agent: Star2Star Media. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER, REFER, UPDATE, NOTIFY, PUBLISH, SUBSCRIBE. Supported: timer, precondition, path, replaces. Content-Length: 0. . U 65.196.170.191:5080 - 65.196.170.129:5060 BYE sip:gw+s...@65.196.170.129:5080;transport=udp SIP/2.0. Via: SIP/2.0/UDP 65.196.170.191:5080;rport;branch=z9hG4bKpv9pFF0S5r9ca. Route: sip:65.196.170.129;lr=on;ftag=3D535BNy2tS9e;did=699.54d14392. Max-Forwards: 70. From: sip:+15673861...@65.196.170.129;tag=p4H9tpXS1Qv8m. To: WIRELESS CALLER sip:+19412848...@65.196.170.129;tag=3D535BNy2tS9e. Call-ID: 2d4c8faa-385d-122d-a69b-003018ae0bd3. CSeq: 121917605 BYE. Contact: sip:+15673861...@65.196.170.191:5080;transport=udp. User-Agent: Star2Star Media. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER, REFER, UPDATE, NOTIFY. Supported: timer, precondition, path, replaces. Reason: Q.850;cause=16;text=NORMAL_CLEARING. Content-Length: 0. . It's bizarre because the re-INVITE never gets transmitted to the other leg and FreeSWITCH responds with an SDP that only supports RFC2833. Here is the FreeSWITCH level 7 log: 2009-10-20 20:54:15.577191 [NOTICE] switch_channel.c:613 New Channel sofia/s2s/+19412848...@65.196.170.129 [ba5b3890-bdba-11de-b004-d7a35c351f6f] 2009-10-20 20:54:15.577191 [DEBUG] switch_core_state_machine.c:306 (sofia/s2s/+19412848...@65.196.170.129) Running State Change CS_NEW 2009-10-20 20:54:15.577191 [DEBUG] sofia.c:3493 Channel
Re: [Freeswitch-users] Proxy media mode with T.38 re-invite
Fix this or set it to silence. /b On Oct 20, 2009, at 3:57 PM, Kristian Kielhofner wrote: EXECUTE sofia/s2s/+19412848...@65.196.170.129 playback (local_stream://moh) 2009-10-20 20:54:19.112927 [ERR] switch_core_file.c:116 Invalid file format [local_stream] for [moh]! ___ 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] 2 voicemail questions
1. Can I email the voicemail message to multiple email addresses?
I revisited this again after not requiring it for a while. A comma
separated list in the extension.xml file does work. The problem was
the template file. Once I removed and in the To: field, I can
send to multple emails no problems:
In 1000.xml (for example)
param name=vm-mailto value=em...@number1.com,em...@number2.com/
And then I editied voicemail.tpl and changed the second line from:
To: ${voicemail_email}
TO:
To: ${voicemail_email}
Not sure if this is a fault in the mail server or not. Maybe the
default templates should be changed to handle this?
On Fri, Jul 10, 2009 at 8:57 PM, Mark Campbell-Smith
mcampbellsm...@gmail.com wrote:
Hi!
1. Can I email the voicemail message to multiple email addresses?
A comma separated list does not work in the extensions.xml file
(1000.xml), but it does work if I hard code the email addresses into
the notify-voicemail.tpl file.
Could this be added to the switch so that it can handle comma separated lists?
2. How can I make Freeswitch dial a number AFTER a voicemail is left?
api Hangup hook?
i want the 'voicemail' application to appear to call the extension to
notify the user that there is a waiting message. This is an extract
from my dialplan.xml:
condition field=destination_number expression=^(10[01][0-9])$
action application=set data=dialed_extension=$1/
action application=export data=dialed_extension=$1/
!-- action application=set data=ringback=${us-ring}/ --
action application=set data=call_timeout=30/
action application=set data=hangup_after_bridge=true/
action application=set data=continue_on_fail=true/
action application=bridge
data=user/${dialed_extensi...@${domain_name}/
action application=answer/
action application=sleep data=1000/
action application=voicemail data=default ${domain_name}
${dialed_extension}/
/condition
condition
field=${vm_boxcount(${destination_numb...@${domain_name})}
expression=^(1)$
action application=log data=MSGBOX
${vm_boxcount(${dialed_extensi...@${domain_name})}/
action application=set data=api_hangup_hook=originate
sofia/internal_nat/${dialed_etension}%${domain_name} default default
Message 4000 4000 3/
This only works if the B leg (ie voicemail application) hangs up
first. This would be an unusual situation and does not achieve what I
want... is there any other way to achieve this?
Thanks
Hi!
I have 2 questions regarding voicemail ...
1. Can I email the voicemail message to multiple email addresses? If
so, what format is this in?
param name=vm-mailto value=m...@myemail.com/
Try a comma sep. list. Not sure if it will work.
2. How can I make Freeswitch dial a number AFTER a voicemail is left?
api Hangup hook?
I g
From: Brian West br...@freeswitch.org
On Jul 7, 2009, at 9:11 AM, Mark Campbell-Smith wrote:
Hi!
I have 2 questions regarding voicemail ...
1. Can I email the voicemail message to multiple email addresses? If
so, what format is this in?
param name=vm-mailto value=m...@myemail.com/
Try a comma sep. list. Not sure if it will work.
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Re: [Freeswitch-users] Can not record session. Media not enabled on channel.
What's in the dialplan for this channel? Is bypass-media or proxy-media set to true? Do a debug trace and post it in pastebin. -MC On Tue, Oct 20, 2009 at 3:30 AM, Maciej Aniserowicz maciej.aniserow...@gmail.com wrote: Hello, I am using the same set of extensions for testing the system during development, they include XLite, Cisco sip phone and several extensions that just play some audio file. Sometimes, very rarely, this message Can not record session. Media not enabled on channel. appears on FS console. Like I wrote before, I don't change any codec settings and always use the same set of devices/emulators. What can cause this message? Thanks, Maciej Aniserowicz ___ 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] CS_REPORTING Channel event state
REPORTING is the state that it writes to CDR. If you have calls stuck in this state, take one and try to use uuid_kill on it and see if it goes away, then get a core off of it and pastebin the thread apply all bt (with no other calls up). What modules are you using for cdr and with what configuration? Mike On Oct 20, 2009, at 11:47 AM, Dome Charoenyost wrote: Dear All What's CS_REPORTING state ? I found many channels not hang up ans state is CS_REPORTING e264f84a-bd87-11de-9a90-2320c02172de,outbound,2009-10-20 21:50:26,1256050226,sofia/external/x...@xxx.xxx.xx. 191:7050,CS_REPORTING,FreeSWITCH,,xx.xxx.xxx.xxx, 7050,,,XML,public, ___ 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] Connect PHP SOAP Web Server with SQLite database of FS
If you really want to access this information outside I would strongly recommend using odbc instead of the internal sqlite db, it does not handle locking contention well. If you need access to things in the core db (like show calls and show channels information) you will need to write a small daemon that listens on events socket and puts that information into a database. Mike On Oct 20, 2009, at 2:23 PM, Chris Burns wrote: If you really wanted: http://php.net/manual/en/book.sqlite.php But I would recommend you make use of ODBC to use a client/server RDBMS. Here's some good reading: http://www.sqlite.org/cvstrac/wiki?p=WhenToUseSqlite On October 20, 2009 10:53:01 am homqua wrote: Now I am building a PHP SOAP Web Service to access the database of FS. Anyone has idea about how to access sqlite database of FS through PHP ? I have read about socket event in FS, but I don't know whether it can response with the query of database or not. Thanks for your help. ___ 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] Proxy media mode with T.38 re-invite
Brian, It's already set to silence :). My guess here is that Sofia is mis-interpreting the re-INVITE as not having any media (a la RFC 2543 hold). I don't want to place the call on hold. I want to negotiate T.38 ;). On Tue, Oct 20, 2009 at 5:35 PM, Brian West br...@freeswitch.org wrote: Fix this or set it to silence. /b On Oct 20, 2009, at 3:57 PM, Kristian Kielhofner wrote: EXECUTE sofia/s2s/+19412848...@65.196.170.129 playback (local_stream://moh) 2009-10-20 20:54:19.112927 [ERR] switch_core_file.c:116 Invalid file format [local_stream] for [moh]! ___ 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
Re: [Freeswitch-users] Proxy media mode with T.38 re-invite
Anthony, I will the next chance I get. Thanks! On Tue, Oct 20, 2009 at 5:34 PM, Anthony Minessale anthony.miness...@gmail.com wrote: issue: console loglevel debug sofia profile internal siptrace on and put it on pastebin http://pastebin.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
Re: [Freeswitch-users] Proxy media mode with T.38 re-invite
Done. On Tue, Oct 20, 2009 at 5:34 PM, Anthony Minessale anthony.miness...@gmail.com wrote: issue: console loglevel debug sofia profile internal siptrace on and put it on pastebin http://pastebin.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
Re: [Freeswitch-users] can't dial from IPv6 to IPv6
ineya ineya ine...@gmail.com wrote:
Hmm, I didn't about this.
so in directory/default.xml I have:
param name=dial-string
value={presence_id=${dialed_us...@${dialed_domain}}${sofia_contact(${dialed_us...@${dialed_domain})}/
and the modified version for IPv6 would be ?...
param name=dial-string
value={presence_id=${dialed_us...@${dialed_domain}}${sofia_contact(internal-ipv6/${dialed_us...@${dialed_domain})}/
Yes. Are you suggesting it didn't work?
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[Freeswitch-users] Hi Guys
Hi to all of you guys (and ladies too). I would like to bother all of you and take just a little of your time. I am a total NOOB on freeswitch and would like a little help. I need to know from all of you guys which would be the hardware recommended from all of you guys to be able to deal with a DS3. The thing is i have a CANTATA switch already with the DS3 and making calls. But i will have to communicate the company international headquarters through there which in turn there are 21 offices around the world and at a certain time all of them will be passing from freeswitch server to the land lines. So could you pleae help out here. Thanks to all of you and the ones who take the time to reply. Ed PS All the calls will be comming from the web through SIP protocol into the freeswitch going out to cell phones and land lines for the executives to handle the calls. Call flow as this. A sip client on the webserver connects to the freeswitch, registers and makes the call to the support agent or the executive. Thats what we are implementing. So i guess the web server has to be another little animal to handle the traffic too. Thanks again. ___ 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
