July 2013 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Make agent unavailable(logged out) automatically
by Sayyed Mohammad Emami Razavi 06 Jul '13

06 Jul '13

Can i make agent unavailable automatically when his/her connection lost for any reason? For example the agent was logged in to the callcenter and after some minutes his/her connection has been broken. I want to set this agent's status unavailable(logged out) automatically.

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Version 1.5.3b: callcenter was not loaded after freeswitch being made!
by Sayyed Mohammad Emami Razavi 06 Jul '13

06 Jul '13

before installation: [root@emami freeswitch]# pwd /usr/local/src/freeswitch [root@emami freeswitch]# vim modules.conf #applications/mod_blacklist applications/mod_callcenter #applications/mod_cidlookup applications/mod_cluechoo applications/mod_commands after installation: freeswitch@internal> status UP 0 years, 0 days, 0 hours, 10 minutes, 25 seconds, 715 milliseconds, 274 microseconds FreeSWITCH (Version 1.5.3b git 9ed53db 2013-07-06 00:50:57Z) is ready 0 session(s) since startup 0 session(s) - 0 out of max 30 per sec 1000 session(s) max min idle cpu 0.00/94.00 Current Stack Size/Max 240K/8192K freeswitch@internal> module_exists mod_callcenter false freeswitch@internal> mod_callcenter reload -ERR mod_callcenter reload Command not found!

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[SOLVED] Version 1.5.3b: callcenter was not loaded after freeswitch being made!
by Sayyed Mohammad Emami Razavi 06 Jul '13

06 Jul '13

*NOTE* I forgot uncomment <load module="mod_callcenter"/> in autoload_configs/modules.conf.xml ---------- Forwarded message ---------- From: Sayyed Mohammad Emami Razavi <emamirazavi(a)gmail.com> Date: Sat, Jul 6, 2013 at 10:35 AM Subject: Version 1.5.3b: callcenter was not loaded after freeswitch being made! To: freeswitch-users(a)lists.freeswitch.org before installation: [root@emami freeswitch]# pwd /usr/local/src/freeswitch [root@emami freeswitch]# vim modules.conf #applications/mod_blacklist applications/mod_callcenter #applications/mod_cidlookup applications/mod_cluechoo applications/mod_commands after installation: freeswitch@internal> status UP 0 years, 0 days, 0 hours, 10 minutes, 25 seconds, 715 milliseconds, 274 microseconds FreeSWITCH (Version 1.5.3b git 9ed53db 2013-07-06 00:50:57Z) is ready 0 session(s) since startup 0 session(s) - 0 out of max 30 per sec 1000 session(s) max min idle cpu 0.00/94.00 Current Stack Size/Max 240K/8192K freeswitch@internal> module_exists mod_callcenter false freeswitch@internal> mod_callcenter reload -ERR mod_callcenter reload Command not found!

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unable to send fax using txfax on LAN
by arun sethi 06 Jul '13

06 Jul '13

I am trying to send fax to a local fax machine (Canon MX 512) on my LAN using freeswitch. The machine is connected via linksys pap2 (ATA) on LAN to freeswitch. the command I am using on freeswitch console is: /usr/local/freeswitch/bin/fs_cli -x "originate {fax_verbose=true,ignore_early_media=true,absolute_codec_string='PCMU'}user/1006 &txfax(/tmp/txfax.tiff)" The machine in question is Canon Mx 512 in fax only mode. Note that I am able to send out fax from this machine and receive it in freeswitch using rxfax. The problem is only in sending fax to this machine using txfax. The complete error log is at paste http://pastebin.freeswitch.org/21151 Here are some relevant snippets from error log 2013-07-04 21:23:09.215751 [DEBUG] mod_sofia.c:243 sofia/internal/ sip:1006@10.2.2.159:8081 SOFIA EXECUTE 2013-07-04 21:23:09.215751 [DEBUG] switch_core_state_machine.c:209 sofia/internal/sip:1006@10.2.2.159:8081 Standard EXECUTE EXECUTE sofia/internal/sip:1006@10.2.2.159:8081 txfax(/tmp/txfax.tiff) 2013-07-04 21:23:09.215751 [DEBUG] mod_spandsp_fax.c:1387 Raw read codec activation Success L16 30000 2013-07-04 21:23:09.215751 [DEBUG] switch_core_codec.c:219 sofia/internal/ sip:1006@10.2.2.159:8081 Push codec L16:70 2013-07-04 21:23:09.215751 [DEBUG] mod_spandsp_fax.c:1403 Raw write codec activation Success L16 2013-07-04 21:23:09.265748 [DEBUG] switch_rtp.c:3706 Correct ip/port confirmed. 2013-07-04 21:23:09.265748 [DEBUG] switch_core_io.c:759 Engaging Read Buffer at 480 bytes vs 160 2013-07-04 21:23:09.475748 [WARNING] mod_sofia.c:1279 Asynchronous PTIME not supported, changing our end from 30 to 20 2013-07-04 21:23:09.475748 [DEBUG] sofia_glue.c:3037 Changing Codec from PCMU@30ms@8000hz to PCMU@20ms@8000hz 2013-07-04 21:23:09.535748 [DEBUG] switch_rtp.c:1873 RE-Starting timer [soft] 160 bytes per 20ms 2013-07-04 21:23:09.535748 [DEBUG] sofia_glue.c:3124 Set Codec sofia/internal/sip:1006@10.2.2.159:8081 PCMU/8000 20 ms 160 samples 64000 bits 2013-07-04 21:23:09.555751 [DEBUG] switch_core_io.c:1365 Engaging Write Buffer at 320 bytes to accommodate 480->320 2013-07-04 21:23:12.095748 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 HDLC signal status is Carrier up (-2) in state T 2013-07-04 21:23:15.095748 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 HDLC signal status is Carrier down (-1) in state T 2013-07-04 21:23:15.155748 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 HDLC signal status is Carrier up (-2) in state T 2013-07-04 21:23:15.195750 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 HDLC signal status is Abort (-8) in state T 2013-07-04 21:23:15.235749 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 HDLC signal status is Abort (-8) in state T 2013-07-04 21:23:15.275748 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 HDLC signal status is Abort (-8) in state T 2013-07-04 21:23:15.315750 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 HDLC signal status is Abort (-8) in state T 2013-07-04 21:23:15.355748 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 HDLC signal status is Abort (-8) in state T 2013-07-04 21:23:15.395748 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 HDLC signal status is Abort (-8) in state T 2013-07-04 21:23:15.635748 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 HDLC signal status is Abort (-8) in state T 2013-07-04 21:23:15.715751 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 HDLC signal status is Abort (-8) in state T 2013-07-04 21:23:15.795749 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 HDLC signal status is Abort (-8) in state T 2013-07-04 21:23:15.915751 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 HDLC signal s **** repeats many times **** 2013-07-04 21:23:47.035750 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 HDLC signal status is Abort (-8) in state T 2013-07-04 21:23:48.115755 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 HDLC signal status is Abort (-8) in state T 2013-07-04 21:23:48.135749 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 HDLC signal status is Carrier down (-1) in state T 2013-07-04 21:23:49.535749 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 T0 expired in state T 2013-07-04 21:23:49.535749 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 Status changing to 'Timed out waiting for initial communication' 2013-07-04 21:23:49.535749 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 Disconnecting 2013-07-04 21:23:49.535749 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 Changing from phase A_CNG to E 2013-07-04 21:23:49.535749 [DEBUG] mod_spandsp_fax.c:291 FLOW FAX Set rx type 0 2013-07-04 21:23:49.535749 [DEBUG] mod_spandsp_fax.c:291 FLOW FAX Set tx type 1 2013-07-04 21:23:49.535749 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 Changing from state T to B 2013-07-04 21:23:50.195748 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 Send complete in phase E, state B 2013-07-04 21:23:50.195748 [DEBUG] mod_spandsp_fax.c:511 ============================================================================== 2013-07-04 21:23:50.195748 [DEBUG] mod_spandsp_fax.c:524 Fax processing not successful - result (2) Timed out waiting for initial communication. 2013-07-04 21:23:50.195748 [DEBUG] mod_spandsp_fax.c:529 Remote station id: 2013-07-04 21:23:50.195748 [DEBUG] mod_spandsp_fax.c:530 Local station id: SpanDSP Fax Ident 2013-07-04 21:23:50.195748 [DEBUG] mod_spandsp_fax.c:531 Pages transferred: 0 2013-07-04 21:23:50.195748 [DEBUG] mod_spandsp_fax.c:533 Total fax pages: 0 2013-07-04 21:23:50.195748 [DEBUG] mod_spandsp_fax.c:534 Image resolution: 0x0 2013-07-04 21:23:50.195748 [DEBUG] mod_spandsp_fax.c:535 Transfer Rate: 14400 2013-07-04 21:23:50.195748 [DEBUG] mod_spandsp_fax.c:537 ECM status off 2013-07-04 21:23:50.195748 [DEBUG] mod_spandsp_fax.c:538 remote country: 2013-07-04 21:23:50.195748 [DEBUG] mod_spandsp_fax.c:539 remote vendor: 2013-07-04 21:23:50.195748 [DEBUG] mod_spandsp_fax.c:540 remote model: 2013-07-04 21:23:50.195748 [DEBUG] mod_spandsp_fax.c:542 ============================================================================== 2013-07-04 21:23:50.195748 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 Changing from state B to CALL_FINISHED 2013-07-04 21:23:50.195748 [DEBUG] mod_spandsp_fax.c:291 FLOW T.30 Changing from phase E to CALL_FINISHED 2013-07-04 21:23:50.195748 [DEBUG] mod_spandsp_fax.c:291 FLOW FAX Set rx type 9 2013-07-04 21:23:50.195748 [DEBUG] mod_spandsp_fax.c:291 FLOW FAX FAX exchange complete 2013-07-04 21:23:50.195748 [DEBUG] mod_spandsp_fax.c:291 FLOW FAX Set tx type 9 2013-07-04 21:23:50.195748 [DEBUG] mod_spandsp_fax.c:291 FLOW FAX FAX exchange complete 2013-07-04 21:23:50.215748 [DEBUG] switch_core_codec.c:244 sofia/internal/ sip:1006@10.2.2.159:8081 Restore previous codec PCMU:0. 2013-07-04 21:23:50.215748 [NOTICE] switch_core_state_machine.c:262 sofia/internal/sip:1006@10.2.2.159:8081 has executed the last dialplan instruction, hanging up.

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Cisco SIP client
by Hans E. Kristiansen 05 Jul '13

05 Jul '13

Dear All, I have a rather simple, but irritating problem. We had a freeswitch expert installing our server, but at some point I have lost the part of the documentation which explained how to change the name of the cisco sip phone we are using. I would have thought the hostname was set through the freeswitch server dhcp service, but it only allocates the hostname (which then becomes the extension number, according to my understanding of the documentation). Makes sense? Thanks, bighans

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ClueCon 2013 - Register Right Away to Increase Your Chances of Winning
by Michael Collins 05 Jul '13

05 Jul '13

ClueCon 2013 - Register Today! At ClueCon we love giving stuff away. Our sponsors have given us many cool toys to give away this year. We also have books from Packt Publishing. To top it all off we will once again be giving away ten Android 4 tablets! You can increase your chances of winning by registering right away. Everyone who registers by Friday July 12 will *triple* their chances of winning, so don't delay! You can register online right now. Just visit the ClueCon <https://www.cluecon.com/2013/> site and click the "Register 2013" link. To book your hotel room online, simply click the "Book Your Room" link. Rooms are going fast so don't wait! Great Presentations: WebRTC, VoIP Security, and More 2013 will be the best ClueCon yet. Check out this year's schedule<https://www.cluecon.com/schedule/>and you'll see some familiar favorites as well as new faces. WebRTC is a hot topic, as is security. We are happy to let you know that Phil Zimmermann from Silent Circle will be with us again. Be sure to attend the VoIP Security meetup on Wednesday evening. Feel free to contact us if you have any questions about this year's event. We look forward to seeing everyone in Chicago next month! -- Michael S Collins ClueCon Team http://www.cluecon.com 877-7-4ACLUE

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configure: error: libuuid development files required
by Iwan Budi Kusnanto 05 Jul '13

05 Jul '13

Hi, I try to build latest head, rev f7fb54d0cd22c361298c772ce025a47f7937631d. When doing : ./configure --with-openssl , i got this error: configure: error: libuuid development files required I use ubuntu 12.04 and have these packages installed: $ dpkg -l | grep uuid ii libghc-uuid-dev 1.2.3-2 create, compare, parse and print Universally Unique Identifiers ii libuuid1 2.20.1-1ubuntu3 Universally Unique ID library ii uuid-dev 2.20.1-1ubuntu3 universally unique id library - headers and static libraries ii uuid-runtime 2.20.1-1ubuntu3 runtime components for the Universally Unique ID library I also succesfully build rev 2908e3acd50c54ec3e7f562bd6b7c097359d6670 from yesterday. Is it a bug in latest commit or do i need another packages to be installed? Thanks -- Iwan Budi Kusnanto

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'*' breaking out of dial plan
by lists＠collisters.com 05 Jul '13

05 Jul '13

Dumb question I'm sure, but why does pressing the asterisk key '*' end my dialplan. For example the sample milliwatt tone on 9197 stops when the asterisk is pressed. I can see this is related to the inbuilt IVR app, but is there a way to disable this so my dial plan application can run to completion? Thanks, Paul C

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Odp: Odp: Specify a time limit on a conference recording?
by Mariusz Czulada 05 Jul '13

05 Jul '13

Something like this should work: sched_api +600 group_name conference conf_xxx recording stop rec_xxx_file.wav so try embed it in your dalplan > You can schedule a task to stop recordng at a given time. > > > Is there a way to specify a time limit on a conference recording?

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Odp: Specify a time limit on a conference recording?
by Mariusz Czulada 05 Jul '13

05 Jul '13

You can schedule a task to stop recordng at a given time. > Is there a way to specify a time limit on a conference recording?

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