March 2013 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Gsmopen weird sound incoming
by Peter Hartmann 04 Mar '13

04 Mar '13

Hi, Thanks so much for Freeswitch and Gsmopen! I'm loving it! I'm having an issue with incoming calls to a Gsmopen endpoint (E160). Sometimes I hear a strange sound before my IVR recording starts. Here are some examples: http://hartmanncomputer.com/2013-03-03-11-10-17_501x.wav http://hartmanncomputer.com/2013-03-03-11-12-02_501x.wav Although the two examples are pretty consistent, the sounds, when they happen are generally not. Sometimes it's just a hiss, then maybe another time it will be more of a screech. I'm using Ubuntu 12.04-amd64. I have done some optimizations for Linux Audio and USB audio. These do seem to have improved the sound quality through the dongle but they haven't solved this weird sound problem on incoming calls. If anyone's interested, my tweaks were; using the lowlatency kernel and adding 'snd-usb-audio nrpacks=1' to my /etc/modules. Has anyone experienced this sound? Thanks much! Peter Hartmann Hartmann Computer Consulting

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NEW tutorial on Realtime OpenSIPS - FreeSWITCH Integration
by Giovanni Maruzzelli 04 Mar '13

04 Mar '13

Ciao VOIPers, it's my pleasure to bring to your attention a new tutorial on realtime integration between OpenSIPS and FreeSWITCH. It's a cut and paste tutorial, so you can test it right away, eg on a virtual machine, and when confident customize it and put it in production. The stack is Debian Squeeze 6.x, OpenSIPS 1.8.x, FreeSWITCH 1.2.x, OpenSIPS-CP as GUI, MySQL as database. You can find the tutorial at URL: http://www.opensips.org/Resources/DocsTutFreeSwitch with all required files. Please let us know what do you think about it, and what other tutorials you would like to read (at the moment I'm thinking at an HA install of FusionPBX+FreeSWITCH+OpenSIPS, but other requests will be taken into account too). See below for a small excerpt of this tutorial: ===== 1.1 Scope This tutorial can be used as a cut and paste complete and working installation. Please follow strictly all the steps, in the order given. This tutorial presents the concept and implementation of a realtime integration of OpenSIPS SIP server and FreeSWITCH media server. OpenSIPS is used a SIP server - users are registering with it, it routes calls, etc - while the purpose of FreeSWITCH is to provide a full set of media services - like voicemail, conference, announcements, etc. It is a realtime integration because both OpenSIPS and FreeSWITCH are provisioned in the same time when comes to user accounts - when creating a new OpenSIPS user, automatically FreeSWITCH will learn about it an provide and configure all necessary media services for it. Both OpenSIPS and FreeSWITCH will be provisioned (for user accounts) via a shared mysql database. All FreeSWITCH functionalities will be available to OpenSIPS users by prefixing "*" (eg: star) to the extension dialed. *1234 will be passed to FreeSWITCH as 1234, while **1234 will be passed to FreeSWITCH as *1234 ________________________________ 1.2 Setup presentation This tutorial can be used as a cut and paste complete and working installation. Please follow strictly all the steps, in the order given. The following services will be offered by FreeSWITCH by this integrated configuration: voicemail - users will get access to their mailbox; authentication will be done by OpenSIPS; while FreeSWITCH will only provide voicemail IVR (with access PIN); conference' - OpenSIPS will detect and forward calls related to conference service (based on prefixes) to FreeSWITCH, which will provide access (pin based) to the conference rooms; all functionalities - OpenSIPS users will prefix * to reach the corresponding extension in FreeSWITCH (*1234 will be passed to FreeSWITCH as 1234, while **1234 will be passed to FreeSWITCH as *1234) ===== ciao for now, -giovanni -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618

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Bridge problem from outbound socket
by Denis Gasparin 04 Mar '13

04 Mar '13

Hi. We call an extension called "netcat" which is configured in this way: <extension name="netcat"> <condition field="context" expression="public"/> <condition field="destination_number" expression="^netcat$"> <action application="log" data="INFO extension netcat"/> <action application="socket" data="127.0.0.1:8085 async full"/> </condition> </extension> The inbound call is correctly routed to a netcat socket listening on localhost:8085. When I get the socket connect, I issue the following commands: connect ... sendmsg call-command: execute execute-app-name: answer ... sendmsg call-command: execute execute-app-name: bridge execute-app-arg: sofia/external/external_number@external_domain The bridge is done correctly but the media stream is not bidirectional: • Aleg can listen Bleg • Bleg can speak but doesn't hear Aleg The strange thing is that if I put on hold and unhold Aleg with: sendmsg call-command: execute execute-app-name: hold .. sendmsg call-command: execute execute-app-name: unhold the media stream is recovered correctly: Aleg and Bleg can speak/listen each other. I attach pastebin for completeness: • Freeswitch log: http://pastebin.freeswitch.org/pastebin.php?dl=20656 • Outbound socket log: http://pastebin.freeswitch.org/pastebin.php?dl=20657 Thank you for your help. Denis Gasparin

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How to submit patches (email function)
by Steven Schoch 04 Mar '13

04 Mar '13

I'm using the <number>@vtext.com SMS gateway, which has this annoying feature of including "[Attachment(s) removed]" if the message has any MIME parts. So I modified the function switch_simple_email() in switch_utils.c so that it sends a plain-text message, with no MIME headers, if there is no "file" attachment, and the "body" does not include a "content-type". Is this change something that you all would consider including in the official Freeswitch release? Is jira the proper way to submit something like this? -- Steve

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Re: [Freeswitch-users] Unable to Recreate Log Files
by Peter Olsson 04 Mar '13

04 Mar '13

It might be lots of different reasons, but in the end it boils down to the OS -there is probably something else (outside FS) holding a lock, which make the rename file call to fail. Make sure to exclude the folder from antivirus, and make sure no other applications are touching the log files. The actual method called from FS is MoveFileEx, which is a standard Win32 api call - if it fails it's because the OS can't move the file, there is nothing inside FS that can change this. /Peter Från: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] För Subhash Skickat: den 4 mars 2013 15:11 Till: FreeSWITCH Users Help Ämne: Re: [Freeswitch-users] Unable to Recreate Log Files But after a few hours of stress test we are facing this scenario. Are there any reasons? On Mar 4, 2013 7:27 PM, "Peter Olsson" <POlsson(a)enghouse.com<mailto:POlsson@enghouse.com>> wrote: My guess would be antivirus or something similar, that's holding a file lock when trying to rename the file. /Peter Från: freeswitch-users-bounces(a)lists.freeswitch.org<mailto:freeswitch-users-bounces@lists.freeswitch.org> [mailto:freeswitch-users-bounces@lists.freeswitch.org<mailto:freeswitch-users-bounces@lists.freeswitch.org>] För Subhash Skickat: den 4 mars 2013 08:42 Till: FreeSWITCH Users Help Ämne: [Freeswitch-users] Unable to Recreate Log Files Hi, I used the latest binaries(Windows Installer),version info is given below and observered that when it is not able to rename the log file it is keep on writing the logs in freeswitch.log file which is crossing the limt set in logconf.xml file. The error seen in the freeswitch log file is [CRIT] mod_logfile.c:164 Error renaming log from C:/Program Files/FreeSWITCH/log/freeswitch.log.4 to C:/Program Files/FreeSWITCH/log/freeswitch.log.5 [No error] And the version info is Freswitch version 1.3.13b git 3c7f8f8 2013-02-20 22:44:51z NOTE : This issue is critical as we are facing it at production systems. Thanks, Subhash. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org<mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org<mailto: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 !DSPAM:5134a8c632761755499571!

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Monitoring Phone Endpoint Deregistration/Losing Connection
by Avi Marcus 04 Mar '13

04 Mar '13

I'd like to monitor if my user's sip phones are disconnected, to let me know there might be a problem in advance. I see there's a CUSTOM event called sofia::unregister. I presume that triggers whenever a phone doesn't re-register in enough time. However, with default registation of 60 minutes (or even 10) I'd like more granularity. I have the endpoints pinging FS, but I don't see an event or that information stored anywhere. How about setting nat-options-ping (or better, all-reg-options-ping since most NATed endpoint don't actually self-identify). Do I get an event if that fails? I see unregister-on-options-fail but that seemed too aggressive when I tried that in the past. What exactly does the less sever mark-dead-on-options-fail do? Does it remove it from the user/$NUMBER endpoint? Does it trigger an event for me? Other suggestions? -Avi

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X-FS-Support Header
by Kevin Mathy 04 Mar '13

04 Mar '13

Hi List, After some researches, I didn't have found how to remove the X-FS-Support of SIP INVITES sent by Freeswitch. *Example : X-FS-Support: update_display,send_info* I've tried modifying "ignore_display_updates" variable, as described here : http://wiki.freeswitch.org/wiki/Variable_ignore_display_updates, and modifying "pass_callee_id" variable (even if I doubt about it...) without any success. Have you got any other idea ? It's important for us to be able to suppress this header, because of our SIP provider SBC, which doesn't support it... Thanks a lot, * Bien cordialement, Best Regards, **Kevin MATHY* * * --

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Re: [Freeswitch-users] Unable to Recreate Log Files
by Peter Olsson 04 Mar '13

04 Mar '13

My guess would be antivirus or something similar, that's holding a file lock when trying to rename the file. /Peter Från: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] För Subhash Skickat: den 4 mars 2013 08:42 Till: FreeSWITCH Users Help Ämne: [Freeswitch-users] Unable to Recreate Log Files Hi, I used the latest binaries(Windows Installer),version info is given below and observered that when it is not able to rename the log file it is keep on writing the logs in freeswitch.log file which is crossing the limt set in logconf.xml file. The error seen in the freeswitch log file is [CRIT] mod_logfile.c:164 Error renaming log from C:/Program Files/FreeSWITCH/log/freeswitch.log.4 to C:/Program Files/FreeSWITCH/log/freeswitch.log.5 [No error] And the version info is Freswitch version 1.3.13b git 3c7f8f8 2013-02-20 22:44:51z NOTE : This issue is critical as we are facing it at production systems. Thanks, Subhash. !DSPAM:51344d7e32763958169474!

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suppressed callerid number
by Tamer Higazi 04 Mar '13

04 Mar '13

Hi people! I am interisted to block all calls that come without callerid or the party has suppressed the number. Any ideas how the xml condition has to look like?! Tamer

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Using mod_snmp for monitoring and stats
by Julian Pawlowski 04 Mar '13

04 Mar '13

Hi all, I was integrating mod_snmp for monitoring and stats purposes. However it seems snmpwalk does not show any of the mentioned details mentioned on the mod_snmp Wiki page, only the normal system stuff. I don't get any errors in the logs so I guess mod_snmp and agentx are just running fine all in all. Am I missing something? Cheers, Julian

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