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Running LUA script asynchronously?
by Yehavi Bourvine 30 Jun '15

30 Jun '15

Hi, I would like to add a small functionality to our PBX: when someone calls an emergency number, in parallel to the call, an Email will be sent to our security personel, and a call will be initiated to the security center telling them that extension XXX called an emergency number. I would like this process to run in parallel to the original call. How can I do it? - I prefer calling a LUA script from the dialplan, but it will stall the user's call; can it be run in parallel? - If not, I'll probably use ESL. What is the best event filter string to get informed only about call initiation? Thanks! __Yehavi:

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FreeSWITCH Week in Review (Master Branch) June 20th-26th
by krice＠freeswitch.org 30 Jun '15

30 Jun '15

New Post on freeswitch.org from Kathleen check it out at http://ift.tt/1NrzneF FreeSWITCH Week in Review (Master Branch) June 20th-26th Hello, again. This passed week in the FreeSWITCH master branch we had 37 commits. There was one feature this week with improvements to play_and_detect_speech to set the current_application_response channel variable. Join us on Wednesdays at 12:00 CT for some more FreeSWITCH fun! And head over to freeswitch.com to learn more about FreeSWITCH support. New features that were added: FS-7720 Improve play_and_detect_speech to set current_application_response channel variable as follows: “USAGE ERROR”: bad application arguments’, “GRAMMAR ERROR”: speech recognizer failed to load grammar, “ASR INIT ERROR”: speech recognizer failed to allocate a session, and “ERROR”: any other errors Improvements in build system, cross platform support, and packaging: FS-7707 Fix build error on CentOS7 FS-7655 Fixed a build error when we have PNG but not YUV FS-7723 Change RPMs to use -ncwait instead of -nc. This will cause the initscript to pause and wait for FS to be ready before continuing. FS-7648 Added a test cases for FS-7724 and FS-7687 FS-7726 Additional configurations for a QA test case FS-7715 Updates to configure and spec files for next development branch and added images to spec file and fixed build/freeswitch.init.redhat since redhat likes to override settings in the script with TAGs in comments The following bugs were squashed: FS-7467 [mod_callcenter] Fixing stuck channels using uuid-standby agents FS-7699 [mod_verto] Fixed for browser compatibility FS-7722 Fixed an issue with record_session including params when creating path FS-7489 [mod_unimrcp] Fixed a TTS Audio Queue Overflow FS-7724 [mod_conference] Fixed a segfault when missing fonts when trying to render banner FS-7519 [mod_av] Fixed a regression in the visual appearance of decode app output FS-7703 Fixed a bug caused by answer_delay being set in the default configurations FS-7679 [mod_verto] Fixed a bug causing one way audio on Chrome when video is enabled and when using a sip without video FS-7729 [mod_verto] Fixed the formatting for IPv6 addresses This passed week in the FreeSWITCH 1.4 branch we had 30 commits merged in from master. Security issues: FS-7708 Fixed docs on enabling cert CN/SAN validation New features that were added: FS-7561 [mod_sofia] Add Perfect Forward Secrecy (DHE PFS) FS-7564 [mod_rayo] Added new algorithms for offering calls to clients FS-7623 [mod_amqp] Allow for custom exchange name and type for producers and fixed param name ordering bug caused by exposing these params FS-7720 Improve play_and_detect_speech to set current_application_response channel variable as follows: “USAGE ERROR”: bad application arguments’, “GRAMMAR ERROR”: speech recognizer failed to load grammar, “ASR INIT ERROR”: speech recognizer failed to allocate a session, and “ERROR”: any other errors FS-7743 [mod_skinny] Updated SKINNY on-hook action to hang up all calls on a device, except those in a short list of call states (or perform a blind transfer) and added a hook after completing the hangup operation to start ringing if there is an inbound call active on the device. Improvements in build system, cross platform support, and packaging: FS-7610 Fixed a gcc5 compilation issue FS-7426 Only disable mod_amqp on Debian Squeeze and Wheezy FS-7297 g729 installer The following bugs were squashed: FS-7582 FS-7432 Fixed missing a=setup parameter from answering SDP FS-7650 [mod_verto] Fixed crash when making a call from a verto user with profile-variables in their user profile FS-7678 Fixed for fail_on_single_reject not working with | bridge FS-7612 Fixed invalid json format for callflow key FS-7621 [mod_shout] Fixed a slow interrupt FS-7432 Fixed missing a=setup parameter from answering SDP FS-7573 Fixed 80bit tag support for zrtp FS-7636 Fixed an issue with transfer_after_bridge and park_after_bridge pre-empting transfers FS-7654 Fixed an issue with eavesdrop audio not working correctly with a mixture of mono and stereo FS-7579 [mod_conference] Fixed a bug not allowing suppression of play-file-done FS-7593 [mod_skinny] Fixed a bug where skinny phones would stomp on each other in database when thundering herd occurs FS-7597 [mod_codec2] Fixed encoded_data_len for MODE 2400, it should be 6 bytes. Also replaced 2550 bps bitrate (obsoleted operation mode) by 2400 FS-7604 [fs_cli] Fixed fs_cli tab completion concurrency issues on newer libedit FS-7258 FS-7571 [mod_xml_cdr] Properly encode xml cdr for post to web server FS-7607 Update URLs to reflect https protocol on freeswitch.org websites and update additional URLs to avoid 301 redirects. FS-7479 Fixed a crash caused by large RTP/PCMA packets and resampling FS-7524 [mod_callcenter] Fixing tiers, level and position should default to 1 instead of 0 FS-7622 [mod_amqp] Make sure to close the connections on destroy. Currently the connection is malloc’d from the module pool, so there is nothing to destroy. FS-7689 [mod_lua] Fixed a bug with lua not loading directory configurations FS-7489 [mod_unimrcp] Fixed a TTS Audio Queue Overflow FS-7467 [mod_callcenter] Fixing stuck channels using uuid-standby agents

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One way audio Remote extensions
by Jackson Fisher - Forward.IT 29 Jun '15

29 Jun '15

Hi All, I am having issues with one way audio on inbound calls with extensions that I have set up at a new remote office to the freeswitch server. Phones are registering over the WAN to the Freeswitch server, outgoing calls work without issue just inbound calls. What would be the best approach to troubleshooting where the issue lies. I believe I have all the correct ports forwarded. Do I need stun? Thanks, Jackson Fisher [Email-signature.jpg] Jackson Fisher | Network Engineer A : Level 1 - Unit 8, 55-57 Lathlain Street, Belconnen ACT 2617 P : (02) 6162 0070 E : Jackson(a)forwardit.com.au<mailto:Jackson@forwardit.com.au> W : http://www.forwardit.com.au<http://www.forwardit.com.au/>

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fs_cli not showing debug logs to console
by Dave Greeko 29 Jun '15

29 Jun '15

Dear All, I'm having problem getting any output to the fs_cli console. Nothing is being displayed to the console. In addition, nothing gets written to log file. Output from fs_cli: freeswitch@internal> console loglevel debug -ERR invalid log level +OK console log level set to DEBUG here is what I have in the parsed configuration (freeswitch.xml.fsxml) <configuration name="console.conf" description="Console Logger"> <mappings> <map name="all" value="console,debug,info,notice,warning,err,crit,alert"/> </mappings> <settings> <param name="loglevel" value="debug"/> </settings> </configuration> Both mod_console and mod_logfile are loaded: freeswitch@internal> module_exists mod_console true freeswitch@internal> module_exists mod_logfile true freeswitch@internal> db_cache status db="hostaddr=127.0.0.1 dbname=freeswitch user=fs password=",type="odbc" Type: PGSQL Last used: 0 Total used: 22 Flags: Unlocked, Detached(0) Creator: src/switch_console.c:255 Last User: src/switch_console.c:255 Regards, Dave

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Unable to run fs_cli (returns Error Connecting [Socket Connection Error])
by Craig Stevenson 29 Jun '15

29 Jun '15

I have an automated script that I have been using to build the 1.6/1.7 (master) branch for FreeSWITCH. Debian = Jessie (8.1); git branch = master. Life was good. But, about a week (or two) ago, I was no longer able to run fs_cli from a shell window on my new builds... craig $ fs_cli [ERROR] fs_cli.c:1610 main() Error Connecting [Socket Connection Error] FreeSWITCH itself is running in the background perfectly fine. I can place calls; and tail -f on the log file shows the same activity that would expect to see on the fs_cli console. The process is is running as user "freeswitch" and all of the standard directories are owned by "freeswitch". And no obvious errors in the build log (but honestly, not sure I'd spot the error in all the text if there was one :-) Any hints on what the issue might be? Thanks. --------------------------------------------------------------- For what it is worth, here are the script commands that I use to build FreeSWITCH: # git freeswitch, bootstrap, configure and make... git clone https://freeswitch.org/stash/scm/fs/freeswitch.git sudo ./bootstrap.sh -j sudo ./configure -C perl -i -pe 's/#formats\/mod_vlc/formats\/mod_vlc/g' modules.conf perl -i -pe 's/#applications\/mod_av/applications\/mod_av/g' modules.conf sudo make sudo make install sudo make cd-sounds-install cd-moh-install samples # add user & group freeswitch sudo useradd --shell /bin/bash --password `perl -e 'print crypt("....", "....")'` freeswitch sudo adduser freeswitch sudo sudo chown -R freeswitch:freeswitch /usr/local/freeswitch/ sudo chmod -R o=rwx /usr/local/freeswitch/ sudo chown -R freeswitch:freeswitch /usr/local/src/freeswitch/ sudo chmod -R o=rwx /usr/local/src/freeswitch/ # set to run at startup sudo ln /usr/local/freeswitch/bin/freeswitch /usr/bin/freeswitch sudo ln /usr/local/freeswitch/bin/fs_cli /usr/bin/fs_cli sudo mkdir /etc/freeswitch sudo ln /usr/local/freeswitch/conf/freeswitch.xml /etc/freeswitch/freeswitch.xml sudo chown freeswitch:daemon /etc/freeswitch sudo chmod ug=rwx,o= /etc/freeswitch sudo mkdir /var/lib/freeswitch sudo chown freeswitch:daemon /var/lib/freeswitch sudo chmod -R ug=rwX,o= /var/lib/freeswitch sudo cp /usr/local/src/freeswitch/debian/freeswitch-sysvinit.freeswitch.default /etc/default/freeswitch sudo chown freeswitch:daemon /etc/default/freeswitch sudo chmod ug=rw,o= /etc/default/freeswitch sudo cp /usr/local/src/freeswitch/debian/freeswitch-sysvinit.freeswitch.init /etc/init.d/freeswitch

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improvement in build system
by covici＠ccs.covici.com 29 Jun '15

29 Jun '15

Hi. I am noticing that if a library is missing in building a module, you don't get the error till the actual compile phase -- should not the config script check and give you the error then -- would save a lot of time, in my opinion. It would have to consult the modules.conf, and check the dependencies -- this is the way most packages do it these days. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici covici(a)ccs.covici.com

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pickup leads to showing own number as caller at pickup-actor
by wiebittewas 29 Jun '15

29 Jun '15

trying to use pickup with following two steps: in the std-extension for local calls the bridge is extended for later pickup <action application="bridge" data="user/$1@${domain_name},pickup/$1"/> in another extension the pickup-code is checked and if it matches, this action take place: <action application="pickup" data="$1"/> well technical pickup works as expected, but the phone of the actor shows itself as the caller-id... on the network I can see a P-Asserted-Identity-Header at the 200-reply after the pickup-invite, which contains the caller-id, and as I understand, this shouldn't be this value, but that of the real caller, who is picked up. unfortunately various tries to either force a value or simply drop this header have failed. any suggestions? regards w.

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Intercept .. the other way ..
by Fred Schulz - privat 29 Jun '15

29 Jun '15

Hey, I'd like to intercept a call. But it shouldn't lead into an answred call. I'd like to push the BLF and after that, my phone just has to ring. That's how it works now: Bob calling Allice Mike pushes BLF button Bob's talking to Mike What i want: Bob calling Allice Mike pushes BLF button Bob calls Mike instead Allice <-- <-- I think i'd like to do a uuid_transfer .. how am I able to invokeit from the DP? Thank you Fred

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MWI : Double Record-Route not supported
by Yannick LE COENT 29 Jun '15

29 Jun '15

Hi, I am using FreeSWITCH 1.4.19 with openSIPS as front-end. OpenSIPS is configured to insert a double Record-Route in SUBSCRIBE before sending it to FreeSWITCH. When FreeSWITCH sends the NOTIFY to openSIPS, there is only a single Route; but it should contains two Route built from the Record-Route. Here are a copy of the messages. The top-down Record-Route has not been copied into the Route of the NOTIFY. SUBSCRIBE sip:7002@ineo-telecom SIP/2.0 Record-Route: <sip:192.168.4.195:24692;r2=on;lr;ftag=4924c77c> Record-Route: <sip:192.168.4.195:24680;r2=on;lr;ftag=4924c77c> Via: SIP/2.0/UDP 192.168.4.195:24692;branch=z9hG4bK3727.3489d7b1.0 Via: SIP/2.0/UDP 192.168.2.138:54530;received=192.168.2.138;branch=z9hG4bK-d8754z-d7205f61d07 c1645-1---d8754z-;rport=54530 Max-Forwards: 69 Contact: <sip:7002@192.168.2.138:54530> To: "Yannick LE COENT1"<sip:7002@ineo-telecom> From: "Yannick LE COENT1"<sip:7002@ineo-telecom>;tag=4924c77c Call-ID: MmZkNTFhM2Q2YTA0MWM5NThkMmNkOWJkZWY5NzEwMzA. CSeq: 2 SUBSCRIBE Expires: 300 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO User-Agent: X-Lite release 1103d stamp 53117 Event: message-summary Content-Length: 0 NOTIFY sip:7002@192.168.2.138:54530 SIP/2.0 Via: SIP/2.0/UDP 192.168.4.195;rport;branch=z9hG4bKHc4DaFSg7j6FB Route: <sip:192.168.4.195:24692>;r2=on;lr;ftag=4924c77c Record-Route: <sip:192.168.4.195:24692>;r2=on;lr;ftag=4924c77c Max-Forwards: 70 From: "Yannick LE COENT1" <sip:7002@ineo-telecom>;tag=HYsX4bK90fkS To: "Yannick LE COENT1" <sip:7002@ineo-telecom>;tag=4924c77c Call-ID: MmZkNTFhM2Q2YTA0MWM5NThkMmNkOWJkZWY5NzEwMzA. CSeq: 776626101 NOTIFY Contact: <sip:7002@192.168.4.195:5060> User-Agent: FreeSWITCH-mod_sofia/1.4.19~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Event: message-summary Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Subscription-State: active;expires=300 Content-Type: application/simple-message-summary Content-Length: 91 Messages-Waiting: yes Message-Account: sip:7002@ineo-telecom Voice-Message: 5/0 (0/0) Thanks for any help, Yannick

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Leg B is left hanging
by Vishal Sharma 29 Jun '15

29 Jun '15

Hi All, I am using ESL to control the call flow. My initial call flow is as following 1. INVITE---> FS 2. 180<--------| 3. 200<--------| 4. ------------>ACK 5. *************|-------------->INVITE 6. *************|<---------------180 7. BYE------->| 8. Retransmissions of BYE 9. *************|<----------------200 OK 10. 200(bye)<---| Now Leg B remains hanging... Shouldn't FS have killed it since legA is dead. legB was generated using bridge command. On Application I get Channel Hangup on 200 bye. In channle hangup . there is no legB uuid hence application neither can disconnect the legB nor it can wait on any events/messages. This issue seems to sofia related. Regards, Vishal Sharma f -- SuperReceptionist is now available on Android mobiles. Track your business on the go with call analytics, recordings, insights and more: Download the app here <https://play.google.com/store/apps/details?id=com.superreceptionist>

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