- FreeSWITCH-users - FreeSWITCH Mailing Lists

bridge with port
by Madovsky 17 Jan '11

17 Jan '11

is it possible to provide the port for a bridge like <action application="bridge" data="sofia/internal/user123@domain:5080"/> ??? for now the answer is "MANDATORY_IE_MISSING". Thanks

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nibblebill no bal action
by Rafqat . 17 Jan '11

17 Jan '11

Hi, I would like leg A and leg B transferred to different extensions when the no balance amount is triggered. I have tried to set a different nobal_action in leg A's session to what I set in the dial string for leg b. But it seems whatever action i put into the dial string gets executed for leg A aswell as leg B. After having a look at the wiki, it seems that this is not currently supported. Can any suggest any work arounds please? Cheers Raf

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Simultaneous Ringing Of Internal Extension And PSTN Number
by Jerry Richards 17 Jan '11

17 Jan '11

Hello All, I would like a configuration that rings both an internal extension and a PSTN number (via an internal PRI), but the PRI immediately answers with PROGRESS_MEDIA (i.e. ringback), which cancels the call to the internal extension. Is there a way to prevent Freeswitch from canceling the internal call? I would like both endpoints to ring until one of them actually answers the call. Here is an example of my bridge statement: bridge({presence_id=2002@192.168.72.144}sofia/internal/sip:2002@192.168.72.22:5060;transport=udp,[lcr_carrier=Carrier / Location 1/INTERNAL PRI TRUNK,lcr_rate=0.00200,origination_caller_id_number=4257402463]openzap/smg_prid/a/4259438201@g1) Thanks, Jerry

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Sharing storage between servers
by Jody Rudolph 17 Jan '11

17 Jan '11

I am curious as to just how far you can take sharing disk storage for the purpose of clustering. Is anyone doing this with the voicemail storage directories? Is it possible to share the SQLite internal databases to avoid resorting to ODBC? I realize that isn't likely, but I have access to some high performance SAN hardware and want to take the most advantage possible. Thanks, Jody Rudolph

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Re: [Freeswitch-users] 4-way and more conference & att_xfer
by Михаил Салтанов 17 Jan '11

17 Jan '11

Can I modify att_xfer limits ?

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4-way and more conference & att_xfer
by Михаил Салтанов 17 Jan '11

17 Jan '11

Hello all I use att_xfer to create 3-way conference from usual call/ How can I add more users? Can I change some variables in source or in config only? Thanks Michail Saltanov

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MOH while running bind_digit_action or bind_meta_app
by Dmitry Sytchev 17 Jan '11

17 Jan '11

Hi! I faced strange problem. After running bind_meta_app or bind_digit_action with att_xfer transferee listens to MOH, but when transferer cancels xfer with #, there are no music on hold for transferee on second run of att_xfer . It doesn't even try to launch playback of MOH. I use local channel in att_xfer to allow attended transfer to arbitrary extensions, so I had to add loopback_bowout/bowout_on_execute=false variables, app_xfer did'nt work as expected at all without them. Could this cause such a strange MOH behavior? first try: 2011-01-17 15:09:34.044500 [DEBUG] switch_rtp.c:3098 RTP RECV DTMF *:884 2011-01-17 15:09:34.264674 [DEBUG] switch_rtp.c:3098 RTP RECV DTMF 7:804 2011-01-17 15:09:34.264674 [DEBUG] switch_ivr_async.c:2866 sofia/local/sytchev2(a)85.114.2.200 Processing meta digit '7' [execute_extension::att_xfer XML features] 2011-01-17 15:09:34.264674 [DEBUG] switch_core_session.c:954 Send signal sofia/local/sytchev2(a)85.114.2.200 [BREAK] 2011-01-17 15:09:34.264674 [DEBUG] switch_core_session.c:709 Send signal sofia/local/sytchev2(a)85.114.2.200 [BREAK] 2011-01-17 15:09:34.404746 [DEBUG] switch_core_session.c:954 Send signal sofia/local/sip:sytchev3@192.168.4.130 [BREAK] 2011-01-17 15:09:34.424718 [DEBUG] switch_core_session.c:709 Send signal sofia/local/sip:sytchev3@192.168.4.130 [BREAK] 2011-01-17 15:09:34.564684 [DEBUG] switch_ivr.c:563 sofia/local/sip:sytchev3@192.168.4.130 Command Execute playback(local_stream://moh) EXECUTE sofia/local/sip:sytchev3@192.168.4.130 playback(local_stream://moh) 2011-01-17 15:09:34.564684 [WARNING] mod_local_stream.c:393 Unknown source moh, trying 'default' 2011-01-17 15:09:34.564684 [DEBUG] mod_local_stream.c:421 Opening Stream [default] 8000hz 2011-01-17 15:09:34.564684 [DEBUG] switch_ivr_play_say.c:1236 Codec Activated L16@8000hz 1 channels 20ms 2011-01-17 15:09:34.564684 [DEBUG] switch_ivr.c:563 sofia/local/sytchev2(a)85.114.2.200 Command Execute execute_extension(att_xfer XML features) EXECUTE sofia/local/sytchev2(a)85.114.2.200 execute_extension(att_xfer XML features) 2011-01-17 15:09:34.565718 [INFO] mod_dialplan_xml.c:331 Processing 8126778008 <sytchev2>->att_xfer in context features Dialplan: sofia/local/sytchev2(a)85.114.2.200 parsing [features->dx] continue=false Dialplan: sofia/local/sytchev2(a)85.114.2.200 Regex (FAIL) [dx] destination_number(att_xfer) =~ /^dx$/ break=on-false Dialplan: sofia/local/sytchev2(a)85.114.2.200 parsing [features->att_xfer] continue=false Dialplan: sofia/local/sytchev2(a)85.114.2.200 Regex (PASS) [att_xfer] destination_number(att_xfer) =~ /^att_xfer$/ break=on-false Dialplan: sofia/local/sytchev2(a)85.114.2.200 Action read(3 11 'tone_stream://%(10000,0,350,440)' digits 30000 #) Dialplan: sofia/local/sytchev2(a)85.114.2.200 Action set(origination_cancel_key=#) Dialplan: sofia/local/sytchev2(a)85.114.2.200 Action att_xfer(loopback/${digits}/route_customer_global) 2011-01-17 15:09:34.565718 [NOTICE] switch_core_session.c:2137 Execute read(3 11 'tone_stream://%(10000,0,350,440)' digits 30000 #) EXECUTE sofia/local/sytchev2(a)85.114.2.200 read(3 11 'tone_stream://%(10000,0,350,440)' digits 30000 #) 2011-01-17 15:09:34.567713 [DEBUG] switch_ivr_play_say.c:1236 Codec Activated L16@8000hz 1 channels 20ms 2011-01-17 15:09:44.564756 [DEBUG] switch_ivr_play_say.c:1573 done playing file Second try: 2011-01-17 15:10:10.643923 [DEBUG] switch_rtp.c:3098 RTP RECV DTMF *:804 2011-01-17 15:10:10.964291 [DEBUG] switch_rtp.c:3098 RTP RECV DTMF 7:804 2011-01-17 15:10:10.964291 [DEBUG] switch_ivr_async.c:2866 sofia/local/sytchev2(a)85.114.2.200 Processing meta digit '7' [execute_extension::att_xfer XML features] 2011-01-17 15:10:10.964291 [DEBUG] switch_core_session.c:954 Send signal sofia/local/sytchev2(a)85.114.2.200 [BREAK] 2011-01-17 15:10:10.964291 [DEBUG] switch_core_session.c:709 Send signal sofia/local/sytchev2(a)85.114.2.200 [BREAK] 2011-01-17 15:10:11.104279 [DEBUG] switch_ivr.c:563 sofia/local/sytchev2(a)85.114.2.200 Command Execute execute_extension(att_xfer XML features) EXECUTE sofia/local/sytchev2(a)85.114.2.200 execute_extension(att_xfer XML features) 2011-01-17 15:10:11.104279 [INFO] mod_dialplan_xml.c:331 Processing 8126778008 <sytchev2>->att_xfer in context features Dialplan: sofia/local/sytchev2(a)85.114.2.200 parsing [features->dx] continue=false Dialplan: sofia/local/sytchev2(a)85.114.2.200 Regex (FAIL) [dx] destination_number(att_xfer) =~ /^dx$/ break=on-false Dialplan: sofia/local/sytchev2(a)85.114.2.200 parsing [features->att_xfer] continue=false Dialplan: sofia/local/sytchev2(a)85.114.2.200 Regex (PASS) [att_xfer] destination_number(att_xfer) =~ /^att_xfer$/ break=on-false Dialplan: sofia/local/sytchev2(a)85.114.2.200 Action read(3 11 'tone_stream://%(10000,0,350,440)' digits 30000 #) Dialplan: sofia/local/sytchev2(a)85.114.2.200 Action set(origination_cancel_key=#) Dialplan: sofia/local/sytchev2(a)85.114.2.200 Action att_xfer(loopback/${digits}/route_customer_global) 2011-01-17 15:10:11.104279 [NOTICE] switch_core_session.c:2137 Execute read(3 11 'tone_stream://%(10000,0,350,440)' digits 30000 #) EXECUTE sofia/local/sytchev2(a)85.114.2.200 read(3 11 'tone_stream://%(10000,0,350,440)' digits 30000 #) 2011-01-17 15:10:11.106278 [DEBUG] switch_ivr_play_say.c:1236 Codec Activated L16@8000hz 1 channels 20ms 2011-01-17 15:10:21.104543 [DEBUG] switch_ivr_play_say.c:1573 done playing file -- Best regards, Dmitry Sytchev, IT Engineer

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Re: [Freeswitch-users] console
by JRichey 17 Jan '11

17 Jan '11

Did you ever try using netstat like Steven suggested? What do you get as output for "netstat -tunlp"? In one of your commands below you show 127.0.0.2 instead of 127.0.0.1 so if that's not a typo you may want to try it again. -Justin -----Original Message----- From: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org]On Behalf Of Sam Sent: Tuesday, January 11, 2011 9:27 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] console Tried this below and received :- # /usr/local/fs_2/bin/fs_cli -H 127.0.0.2 -P 8082 [ERROR] libs/esl/fs_cli.c:1206 main() Error Connecting [Socket Connection Error] tried by creating profile:- # /usr/local/fs_2/bin/fs_cli profile1 [ERROR] libs/esl/fs_cli.c:1206 main() Error Connecting [Socket Connection Error] By port: # /usr/local/fs_2/bin/fs_cli -P 8082 [ERROR] libs/esl/fs_cli.c:1206 main() Error Connecting [Socket Connection Error] netstat -nlp | grep ; ports 8081 & 8082 are available But i could get a console for other server by # /usr/local/fs_2/bin/fs_cli Regards Sam On Tue, Jan 11, 2011 at 10:54 PM, Steven Ayre < steveayre(a)gmail.com <mailto:steveayre@gmail.com> > wrote: Actually, correction it does - just not the event_socket.conf.xml one. It'll read .fs_cli_conf in your home directory if it exists, but that isn't created by default - you create it yourself if you want to use it (it's optional). The command line arguments -H and -P -will- override the config file though. Are you using a capital P? -p is password while -P is port. If there's no password on the event socket you'd get no error from using a small p by accident. -Steve On 11 January 2011 16:59, Steven Ayre < steveayre(a)gmail.com <mailto:steveayre@gmail.com> > wrote: It does not read any config file. On 11 January 2011 16:56, Sam < u2nsam(a)gmail.com <mailto:u2nsam@gmail.com> > wrote: the port and ip donot work for me, is it that the fs_cli is not reading the config from 192.168.2.2 but it is reading the config only of 192.168.2.1, though its in the different [FS_1 & FS_2] path where i am executing. Regds Sam On Tue, Jan 11, 2011 at 9:38 PM, Steven Ayre < steveayre(a)gmail.com <mailto:steveayre@gmail.com> > wrote: /usr/local/FS_1/bin/fs_cli -P 8021 /usr/local/FS_2/bin/fs_cli -P 8022 fs_cll doesn't read any config file. It's not part of the FS server at all, you can have it on a different machine that doesn't have FS installed. It entirely relies on the arguments to control where to connect to. -Steve On 11 January 2011 15:04, Sam < u2nsam(a)gmail.com <mailto:u2nsam@gmail.com> > wrote: Something more here ... i am getting the console for 192.168.2.1 every time i do fs_cli on both instances . like /usr/local/FS_1/bin/fs_cli /usr/localFS_2/bin/fs_cli i get the console for the 1st server only the 2 server are listing to 2 different ips . Regds Sam On Tue, Jan 11, 2011 at 8:07 PM, Steven Ayre < steveayre(a)gmail.com <mailto:steveayre@gmail.com> > wrote: It should work. Is there anything already listening on port 8022? $ netstat -a -n -p | grep 8022 Are you also sure that they're not both loading the same config file? Regards, -Steve On 11 January 2011 14:32, Sam < u2nsam(a)gmail.com <mailto:u2nsam@gmail.com> > wrote: > the scenario is i have 2 ips on 1 server for 2 FS instances; > > 192.168.2.1 > 192.168.2.2 > > and the parameters i have set is:- > > for > 192.168.2.1:- > <param name="listen-ip" value="127.0.0.1"/> > <param name="listen-port" value="8021"/> > for > 192.168.2.2:- > <param name="listen-ip" value="127.0.0.1"/> > <param name="listen-port" value="8022"/> > > Ideally it should work but i am getting console for only 192.168.2.1 FS . > > > Regards > Sam > > On Tue, Jan 11, 2011 at 4:20 PM, Steven Ayre < steveayre(a)gmail.com <mailto:steveayre@gmail.com> > wrote: >> >> <param name="listen-ip" value="127.0.0.1"/> >> <param name="listen-port" value="8021"/> >> >> You can bind both to port 8021 on their individual IPs, or different >> ports on the same IP. >> >> A listen IP of 0.0.0.0 will mean any IP. >> >> -Steve >> >> On 11 January 2011 10:44, Sam < u2nsam(a)gmail.com <mailto:u2nsam@gmail.com> > wrote: >> > A query, >> > >> > I have 2 FS running on one server on 2 different ips, >> > so when i do fs_cli going to respective bins , i see console of only the >> > first server. >> > >> > Is there any way to get the console of both the FS on the same server . >> > I tried changing the port of event socket to 8022 but it donot works. >> > <param name="listen-port" value="8022"/> >> > >> > Is there some method to start the console of both the instances. >> > >> > Regds >> > Sam >> > >> > _______________________________________________ >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users(a)lists.freeswitch.org <mailto:FreeSWITCH-users@lists.freeswitch.org> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> >> > UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users> >> > http://www.freeswitch.org <http://www.freeswitch.org> >> > >> > >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users(a)lists.freeswitch.org <mailto:FreeSWITCH-users@lists.freeswitch.org> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> >> UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users> >> http://www.freeswitch.org <http://www.freeswitch.org> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org <mailto:FreeSWITCH-users@lists.freeswitch.org> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> > UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users 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Find audio problem
by Jonas Gauffin 17 Jan '11

17 Jan '11

Hello, I got two different customers which have problems with that the audio stops working in one direction after a while. And not for all calls but only some. person calling -> My GW provider -> internet -> freeswitch -> internet -> router -> customer. It's always the customer who can't hear the person calling. Any suggestions on how I can find and fix the problem? Thanks, Jonas

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read conference PIN from a file or db
by deniro 17 Jan '11

17 Jan '11

Hi Regarding providing PIN numbers to a conference how can I read PINs from a file or database and allow them to go to a meeting thx deniro--

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