May 2014 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Re: [Freeswitch-users] Custom variables for mod_conference
by Phil Dunks 28 May '14

28 May '14

Hi Michael Thanks for looking at this - I have submitted a patch in FS-6559. Let me explain the reason that the feature would be useful to us. We have an application layer that provides customer specific conference features via mod_conference. We wish to further develop this to provide potential features such as : Questions and Answers : guests in lecture mode press some dtmf to indicate they wish to ask a question, and the chair presses some dtmf to manage the queue and unmute the questioner. Chair greeting : the chair can record a personal greeting message which is played to guests when they join. Security Passcode : the chair can dynamically set a further PIN on the conference instance via DTMF. Chair privacy : a chair can invoke privacy via some DTMF so all guests are put on hold, and only chairs can speak. And others. All of these features will require us to store state relevant to the conference instance. It would make life a lot easier if we could store this state as custom variables on the conference object via the mod_conference api. Best Regards Phil > > > From: Michael Jerris <mike(a)jerris.com> > Subject: Re: [Freeswitch-users] Custom variables for mod_conference > Date: 27 May 2014 18:22:15 BST > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Reply-To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > > > we are happy to look at the patch, just submit it on http://jira.freeswitch.org. That being said, conferences do not have their own variables, so I don't get what this feature is doing. > > > On May 27, 2014, at 3:10 PM, Phil Dunks <phil.dunks(a)netdev.co.uk> wrote: > >> Hi FS Guys >> >> It would be very useful for our application if we could set some arbitrary custom variables into a conference via the mod_conference api (e.g similar to uuid_setvar for setting channel variables). >> >> I have updated mod_conference to allow this, and it is working ok for me. >> >> I’m certainly not a C expert, but basically copied the pattern for other api commands. >> >> If I submit it as a patch please would you consider it for inclusion (and make any corrections that you feel necessary). >> >> Thanks >> >> Phil >> >>

2 1

CallerID withheld problem
by Alex Lake 28 May '14

28 May '14

Hello there.... One of our carriers has changed their signalling today and its resulted in some strange behaviour through our Freeswitch box. The change was that the P-Asserted ID has changed format to have a "+44" at the beginning - it used to be in the national format (eg. "02070601234"). The problem manifests itself further down the chain (the FS box forwards to an Asterisk box which forwards to a legacy system using E1's and Aculab cards). For some reason the E1 interprets the signalling as withheld caller id. Message Header is: Max-Forwards: 69 Session-Expires: 3600;refresher=uac Min-SE: 600 Supported: timer, 100rel To: <sip:448700802332@213.239.197.41> From: <sip:02070604380@217.145.67.2>;tag=3609589914-358461 P-Asserted-Identity: <sip:+442070604380@193.113.183.202> Call-ID: 3560925-3609589914-358457(a)MSX1.mydomain.com CSeq: 1 INVITE Allow: CANCEL, ACK, INVITE, BYE, OPTIONS, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE, MESSAGE, PUBLISH Via: SIP/2.0/UDP 217.145.67.2:5060;branch=z9hG4bK630bccab557da527f8875b902c88fb6b Contact: <sip:02070604380@217.145.67.2:5060> Content-Type: application/sdp Accept: application/sdp Content-Length: 373 I forward this on as: Via: SIP/2.0/UDP 213.239.197.41;rport;branch=z9hG4bKm3jHajt7g1gaa Max-Forwards: 68 From: "02070604380" <sip:02070604380@213.239.197.41>;tag=57cmagKg0yctN To: <sip:448700802332@89.202.128.200> Call-ID: 81dbf634-5ad9-1232-1f90-d43d7ee2fdec CSeq: 59964493 INVITE Contact: <sip:mod_sofia@213.239.197.41:5060> User-Agent: FreeSWITCH-mod_sofia/1.2.23+git~20140319T002132Z~b96946822d~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Allow-Events: talk, hold, conference, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Privacy: none Content-Type: application/sdp Content-Disposition: session Content-Length: 207 X-FS-Support: update_display,send_info P-Asserted-Identity: "02070604380" <sip:02070604380@213.239.197.41> I don't know if the SIPerati can discern anything from this. I could pastebin the pcap file if that would be useful. But the angle might be for me to tell FS to give calls a sip_cid_type of "none". But I don't quite undersrand from the wiki (https://wiki.freeswitch.org/wiki/Variable_sip_cid_type) where I should put this. The current dialplan is: <extension name="ajl-test2"> <condition field="destination_number" expression="^44"> <action application="privacy" data="no"/> <action application="log" data="INFO caller_id_number 0${caller_id_number:3}"/> <action application="export" data="origination_caller_id_number=0${caller_id_number:3}"/> <action application="bridge" data="sofia/internal/${destination_number}(a)89.202.128.200"/> </condition> </extension> Note the hack to change callerid from +44xxx... format to 0xxx... has been replaced by a Lua script that handles it a bit more cleanly. But that "privacy" setting doesn't seem to be enough. Any ideas or suggestions for more diagnostics?

3 6

Backups outbound calls via several isp
by Roman 28 May '14

28 May '14

Francis, thx! But if each isp use custom effective_caller_id_number format and destination_number rule? -- Kind regards, Roman

2 4

Changing bitrate of conference recording file
by Karol Matuszewski 28 May '14

28 May '14

Hi, I see the default values in conference.conf.xml allow to create a recording wav file with parameters: PCM 8000Hz mono 128Kbps I can modify the file using codec after recording. Is it possible to change the value of bitrate before I start recording, using some parameter? -- Regards, Karol Matuszewski

1 0

FreeSWITCH Training and more at ClueCon 2014 Update! Have you registered?
by Ken Rice 27 May '14

27 May '14

Come for the Conference, Stay for the training! The ClueCon Crew is happy to announce FreeSWITCH Training will happen this year on Friday Aug 8th! This will be a full day of training! The following topics will be covered: * FreeSWITCH Basics * Setting up and using WebRTC * Security (SRTP and ZRTP) and Certificates * Using FreeSWITCH¹s ³other² configs FreeSWITCH as a proxy or hybrid * Multi-tenancy with FreeSWITCH * XML/Curl and controlling FreeSWITCH real time The extra day of training is $299 and space is limited! Get your registration in today! Don¹t Forget! Those that register by Friday May 30th will receive those extra entries into the Drawing! After the 30th your chances to win are cut in half! Register Today! http://www.cluecon.com/ -- http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH

1 0

FreeSWITCH Week in Review(Master Branch) May 18th-24th
by Kathleen King 27 May '14

27 May '14

Hello, again. This passed week in the FreeSWITCH master branch we had 37 commits. We saw the addition of a new module, mod_graylog2, which when asked about the module the author, Chris Rienzo, said "Basically, mod_graylog2 allows FS to send logs to a graylog2 server in its native log format: http://graylog2.org/gelf . This GELF implementation is simplified in that I didn't implement compression or chunking, though you still get logs up to around 8192 bytes (minus any fields and JSON overhead). This is an improvement over the 1024 byte limit of syslog. If anybody is interested in adding compression or chunking, feel free to send patches! In addition to supporting larger logs, we are also able to structure the logs such that fields like call UUIDs are easily searchable without having to write a bunch of extractor regexes. We're not yet sure how well graylog2 will scale for us, but the logging code was simple enough to implement in an afternoon that it's worth a try." The following bugs were squashed: ea78f4d Fix race in erlang process spawning bedf832 FS-6418 FS-6427 add profile param NDLB-proxy-never-patch-reinvites to solve both issues Jira: http://jira.freeswitch.org/browse/FS-6418 1ca1d63 FS-6524,FS-6494: the compiler for rpi seems to be strangely mis-optimizing this if/else block. Yes, its already set to this value, but without the else block, it 'magically' hits the if blocks above, unless you sleep, print a log, or add the else Jira: http://jira.freeswitch.org/browse/FS-6524 f683ac2 FS-6533: --resolve fix gcc 4.9 warning due to useless right-hand operand of comma expression in sofia-sip Jira: http://jira.freeswitch.org/browse/FS-6533 New features that were added: b08138d FS-4296 --resolve initial calling/called nums in \'show channels\. Patch provided by Mariusz Czu?ada <manieq.net(a)gmail.com> Jira: http://jira.freeswitch.org/browse/FS-4296 b5816d3 added mod_graylog2: native GELF logging 054c645 FS-4296 --resolve Show initial channel values with 'show channels'. Thanks-to: Mariusz Czu?ada <manieq.net(a)gmail.com> 55f8f55 add heartbeat_fire_on_set variable to fire heartbeat instantly when its first enabled (legacy behavior) in FS core Improvements in cross platform build supports:757b744 Compile and enable OPUS by default to lower the number of steps to complete a WebRTC setup f8527f5 FS-6505 VS2010 Express Jira: http://jira.freeswitch.org/browse/FS-6505 285e7dc slight tweak to pgsql version detection using awk instead of sed for bsd portability 9d99c34 Break on missing ldns lib at configure time af4380d fix linker on mod_cepstral to work in newer environments 1fe6fb6 Use CPPFLAGS for mod_java includes dc671d9 fix name of fs_ivrd c7f2a19 FS-6505 download custom ldns lib for windows for now Jira: http://jira.freeswitch.org/browse/FS-6505 e6ac87f fixes for VS2010 code analysis - hope these are OK else-where In terms of stability these were the use cases that were fixed: e6e0deb Fix erlang deadlock in mod_erlang_event Performance enhancements: 042b162 back off locking contention on sql queues when they are nearly full to allow the system to recover from an attack Feedback is welcome at kathleen.king(a)quentustech.com and the commits for this week are in the attached file along with the corresponding Jira links. -- Kathleen King Quentus Technologies, INC 1037 NE 65th St Suite 273 Seattle, WA 98115 Main: (877) 211-9337 Cell: (703) 859-3757 kathleen.king(a)quentustech.com

1 0

problem with wss
by Jamie McClelland 27 May '14

27 May '14

Hi all - I'm very excited to see support for websockets in 1.4.4! However, I'm having trouble getting wss working. When I enable it, by adding the following to internal.xml: <param name="wss-binding" value=":7443"/> I get the following error when I reload mod_sofia: 2014-05-27 09:24:47.910948 [ERR] sofia.c:2778 Error Creating SIP UA for profile: internal (sip:mod_sofia@209.234.253.18:5060;transport=udp,tcp) The likely causes for this are: 1) Another application is already listening on the specified address. 2) The IP the profile is attempting to bind to is not local to this system. When I comment out the wss line, the error goes away. Strange... 209.234.253.18 is local to the machine, and I'm specifying 7443 as the wss bind IP, not 5060 (which is specified as the "sip_port" in the internal.xml file). If I change the sip_port to a different number, I get the error, but it specifies the new sip_port. Also, accordig to lsof, port 7443 is not in use. I have ws enabled without problems. Here are the only differences between my sip_profile directory and the vanilla conf one: 0 paul:/etc/freeswitch# diff -ur /usr/share/freeswitch/conf/vanilla/sip_profiles/ /etc/freeswitch/sip_profiles/|less Only in /etc/freeswitch/sip_profiles/external: mfpl-voicepulse.xml diff -ur /usr/share/freeswitch/conf/vanilla/sip_profiles/internal.xml /etc/freeswitch/sip_profiles/internal.xml --- /usr/share/freeswitch/conf/vanilla/sip_profiles/internal.xml 2014-05-22 14:39:26.000000000 -0400 +++ /etc/freeswitch/sip_profiles/internal.xml 2014-05-27 09:29:29.000000000 -0400 @@ -196,7 +196,7 @@  <param name="tls-sip-port" value="$${internal_tls_port}"/>  -  + <param name="tls-cert-dir" value="/etc/freeswitch/certs"/>  <param name="tls-passphrase" value=""/>  @@ -304,7 +304,7 @@  -  + <param name="ws-binding" value=":5066"/>   Any suggestions? jamie p.s. I do have wss.pem in /etc/freeswitch/certs -- May First/People Link Growing networks to build a just world http://www.mayfirst.org https://support.mayfirst.org OpenPGP Key: http://current.workingdirectory.net/pages/identity/ xmpp: jamie(a)im.mayfirst.org

5 10

Flowroute is testing CNAM lookup
by Steven Schoch 27 May '14

27 May '14

I am one of the Beta testers of Flowroute's CNAM lookup for inbound calls. So far, it seems to be working quite well. -- Steve

3 4

Voicemails chopped at 10 seconds (randomly!)
by Peter Hartmann 27 May '14

27 May '14

Hey, I have a user that randomly gets voicemails chopped off at exactly 10 seconds. This is using a Sangoma A200 analog card....so it's not a NAT problem. Apparently this has nothing to do with silence detection as I've tried different settings for that. Cutoffs happen in mid word and the audio file length is exactly the same in each when it happens....not always. There's nothing unique about the way and which calls get to this users mailbox. IVR does this: "transfer 1201 XML default". Users mailbox is 1201 and the caller is redacted to 212xxxxxxx. Here's the FS Log for one of the calls that had a 10 second voicemail. http://pastebin.com/Fq5N9UCq Here's Local Extension dialplan snippet that the calls pass through in the event I'm doing something wrong there. http://pastebin.com/FK0zXs4A Admittedly this is a bit out of date: FreeSWITCH Version 1.5.8b+git~20140124T004436Z~bd7182c390~64bit (git bd7182c 2014-01-24 00:44:36Z 64bit) I'm perplexed! Thank very much in advance! Peter Hartmann Hartmann Computer Consulting http://blog.hartmanncomputer.com (212)203-8870 If I can't explain it to you in plain language, that means I don't understand it.

3 4

Docs Call! And other Reminders!
by Ken Rice 27 May '14

27 May '14

Greetings FreeSWITCHers! Hope all of you in the US had a nice long weekend, but its time to get back to work. Calling all Documenters! Today, Tuesday the 27th of May 2PM CST (1700 GMT) on 888 the weekly Docs Team call will kick off! Join us and help continue the great work! Tomorrow, Wed the 28h of May, Join us as Eric Crist, Dir of IT at ClaimLynx joins us to talk about Faxing and FreeSWITCH on ClueCon Weekly! And Friday May 30th, mark your calendars for the FreeSWITCH Friday BugHunt starting about 2PM Central! Where are all these calls? sip:888@conference.freeswitch.org of course, and if you need PSTN or another method of access see http://wiki.freeswitch.org/wiki/Weekly_Conference_Call_Calling_Instructions -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH

2 2