- FreeSWITCH-users - FreeSWITCH Mailing Lists

2 B Channel Transfer (2BCT) and FreeSWITCH
by Doug Albrechtsen 07 May '10

07 May '10

Is anyone having success using 2B Channel Transfer with FreeSWITCH and Sangoma? If so, what are the driver considerations and what does it take to make the transfer work? Thanks, Doug

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Registration ODBC feeded by another registrar proxy
by Rod. 07 May '10

07 May '10

Hi list, was playing with FS 1.0.6 and trying to test the registration performance of FS. (Yes I know FS is more suited as a B2BUA, but please read further :p) So I did the following: - generate one xml file with 20 000 user account like this: <include> <user id="1"> <params> <param name="password" value="1234"/> </params> </user> <user id="2"> <params> <param name="password" value="1234"/> </params> ... Then I used Sipp to test how many registration per second could be fired to the server (quad core 2.83Ghz). I setup ulimit variables, and disable nat. I got this: - using SQL Lite: unable to get higher than 80 registrations per second (in fact it's less than this number but didn't test too much this setup), I see a lot of retransmission in Sipp - using SQL Lite in ramdisk (tmpfs): OK with 80 registrations per second but not much - using ODBC and mysql: 130 reg/sec is OK With ODBC, above 150 reg/sec I see that FS is stalled to 100-110% CPU, I think it's because I'm using only one SIP profile and that SOFIA is monothreaded for this SIP profile. If I'd like to register every 60sec, the server has to support at least more than 300 registration per second. So I'm wondering if I could setup something like this: - use another SIP Proxy as a registrar and feed the ODBC "sip_registration database" of FS - FS will be able to use this database to setup a call - use FS as the outbound proxy for call routing But what about the user params that have been setup in the xml file above. I think that FS loads the user params each time a user is registered. Comments and advices are welcome. regards, rod.

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4x T1 PRI Gateway
by Max Clark 07 May '10

07 May '10

Hello, I am looking for a 4 port T1/PRI to SIP gateway to interconnect a legacy PBX to a SIP trunk. After spending some time with Google I have come up with the following list of devices that fit the required number of T1 interfaces: Dialogic DMG2120DTI Cisco AS5350 Quintum Tenor DX Patton SmartNode 4960 Audiocodes Mediant 1000/2000 Any opinions, personal experiences, advice, or additional devices I should be looking at would be greatly appreciated. Best, Max

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How to turn off SW DTMF detect (for DTMF talkoff issue)
by Robert Hadley 07 May '10

07 May '10

Our Freeswitch implementation has two second audio dropouts on calls that pass through the Sangoma PRI card caused by DTMF talkoff. We know how to turn HW DTMF detection off, but then FS automatically enables software DTMF detect. How do we prevent SW DTMF detection in FS? Our attempt to manage when HW DTMF detection is active is not very clean (see pastebin report of things we tried at http://pastebin.freeswitch.org/12923) I tried using the start_dtmf and stop_dtmf applications in the dialplan but they didn't work to prevent talkoff or splits, and the stop_dtmf application stopped audio for internal calls. Is there a better way to control HW DTMF detection or DTMF line splits from the dialplan? Thanks, Bob Robert

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hanguphook arguments
by Christian Damianidis 07 May '10

07 May '10

Looking at http://wiki.freeswitch.org/wiki/Example_Hangup_hook I see there are some default arguments, however I want to pass in my own. This may be some weak javascript knowledge on my part, but when I do session.setHangupHook(on_hangup(myparam)) It executes the function (obviously as the brackets make it a function call). Any tips on how to do this, is it possible?

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Re: [Freeswitch-users] rfc2833 confusion
by Madovsky 06 May '10

06 May '10

I anyone knows a link of a programming example of how to inject rfc2833 in RTP flow (doesn't matter the language) It would be very useful Thanks F ----- Original Message ----- From: Madovsky To: freeswitch-users(a)lists.freeswitch.org Sent: Thursday, May 06, 2010 12:39 PM Subject: Re: [Freeswitch-users] rfc2833 confusion god, so many confusion for only an audio tone ! ;) anyway if I understand it's called inband by the fact that it's in RTP flow onlly. so the rfc2833 packet is only an event payload ... ----- Original Message ----- From: Anthony Minessale To: freeswitch-users(a)lists.freeswitch.org Sent: Thursday, May 06, 2010 11:52 AM Subject: Re: [Freeswitch-users] rfc2833 confusion both 2833 and actual tones in the audio stream are often called in-band because the 2833 is just a transport for the inband tones that is assumed to be reconstructed on the other side. Some people refer to inband as the real tones and 2833 as not inband because it's not the tones. So the one you have to look out for is people who call 2833 inband and do not clarify that it's inband packetized according the RFC2833 The one that is not inband for sure is INFO (which has been dropped from the standard in favor of some ridiculous SUBSCRIBE based xml exchange) Like all things in SIP and anything else that comes with an RFC even if you know the answer you can't rely on trusting that answer because the real world will do whatever it wants and invalidate you. =D On Thu, May 6, 2010 at 10:36 AM, Madovsky <infos(a)madovsky.org> wrote: HI, is rfc2833 inband or outband ? is it only event or mixed event and sound ? I'm confused after read articles on this subject Thx F _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900 ---------------------------------------------------------------------------- _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org

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rfc2833 confusion
by Madovsky 06 May '10

06 May '10

HI, is rfc2833 inband or outband ? is it only event or mixed event and sound ? I'm confused after read articles on this subject Thx F

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alpha-numeric password
by Mark Himsley 06 May '10

06 May '10

Hi, I'm very new to Freeswitch - I want to move from Asterisk to Freeswitch for my home VoIP switch. In Asterisk sip.conf I have defined extensions like this [201] type=friend username=201 secret=mypass <...SNIP...> If I create ${FREESWITCH_CONF}/directory/default/201.xml starting like this: <include> <user id="201"> <params> <param name="password" value="mypass"/> <...SNIP...> and just change the server the phone connects to to be my new freeswitch server then the phone cannot authentcate: 2010-05-04 11:48:28.936358 [WARNING] sofia_reg.c:1030 SIP auth failure (REGISTER) on sofia profile 'internal' for [201(a)10.0.0.10] from ip 10.0.0.228 But if I set the value element of the password param to a number and also change the password in the phone to that number then the phone will authenticate. Does freeswitch only allow numeric passwords for SIP phones? I can't see a definition on the freeswitch web site for what is valid for a password, and all my Googling has failed too :-( I'd _like_ to keep the passwords unchanged, but I can live with changing them if needed. I was just wondering if I missed something on the freeswitch web site. Thanks in advance. -- Mark

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FreeSWITCH 1.2 (Dragon) on the Horizon Branch Maintainers Wanted
by Anthony Minessale 06 May '10

06 May '10

This month marks 2 years since the release of FreeSWITCH 1.0 (Phoenix) We have added countless features during that time and have held back from updating some or dependancies for stability concerns. The time has come to advance FreeSWITCH to the 1.2 version (Dragon) We would like to branch 1.0 to preserve it but we need volunteers to maintain it. Please contact consulting(a)freeswitch.org or bug us on IRC if you are interested in helping to maintain the 1.0 branch. -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale(a)hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale(a)gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888(a)conference.freeswitch.org> googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888(a)conference.freeswitch.org> pstn:+19193869900

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Can I configure sofia to send an ESL event when REGISTER auth *fails* ?
by david varnes 06 May '10

06 May '10

Hi list, I have listened for "all" events, and the custom "sofia:register" events, but I do not seem to get any event when an auth REGISTER fails. I am trying to get all the info (username, ip address, user-agent etc) that is in the REGISTER event, but for the failed case as well. Thanks for any pointers, davidv -- david varnes e: david.varnes(a)gmail.com p: +61 404 925 633

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