March 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

How do I "register" a domain for a hard phone?
by Jim 29 Mar '10

29 Mar '10

I have just installed Freeswitch and have spent the day searching the wiki for an answer. My soft phone on Linux and the Grand Stream hard phone are finally talking to the switch. However the Grand Stream (on extension 101) requires the soft phone to call it as 101(a)192.168.2.99 whereas the Grand Stream can call the softphone at extension 1000 without an extension. The soft phone is not on the same computer as the switch. The configuration files are identical except for the extensions and passwords. I was getting a message like: 2010-03-28 13:46:52.514487 [WARNING] sofia_reg.c:1872 Can't find user [101(a)192.168.2.51] You must define a domain called '192.168.2.51' in your directory and add a user with the id="101" attribute and you must configure your device to use the proper domain in it's authentication credentials. But I have no idea how to either correct this problem or set a domain anywhere. There is no mention of domain in the configuration for the Grand Stream and changing the user to 101(a)192.168.2.51 in the phone and in the config didn't work. Here are the respective config files: <include> <user id="1000"> <params> <param name="password" value="1234"/> <param name="vm-password" value="1234"/> </params> <variables> <variable name="toll_allow" value="domestic,international,local"/> <variable name="accountcode" value="1000"/> <variable name="user_context" value="default"/> <variable name="effective_caller_id_name" value="Extension 1000"/> <variable name="effective_caller_id_number" value="1000"/> <variable name="outbound_caller_id_name" value="$${outbound_caller_name}"/> <variable name="outbound_caller_id_number" value="$${outbound_caller_id}"/> <variable name="callgroup" value="techsupport"/> </variables> </user> </include> <include> <user id="101" mailbox="101"> <params> <param name="password" value="secret"/> <param name="vm-password" value="1234"/> </params> <variables> <variable name="toll_allow" value="domestic,international,local"/> <variable name="accountcode" value="101"/> <variable name="user_context" value="default"/> <variable name="effective_caller_id_name" value="Extension 101"/> <variable name="effective_caller_id_number" value="101"/> <variable name="outbound_caller_id_name" value="$${outbound_caller_name}"/> <variable name="outbound_caller_id_number" value="$${outbound_caller_id}"/> <variable name="callgroup" value="techsupport"/> </variables> </user> </include> Thanks for any assistance. Jim.

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Experience with FreeSWITCH & Turnserver.org ?
by Fabio Pietrosanti (naif) 29 Mar '10

29 Mar '10

Hi, does anyhave had experience in using FreeSWITCH along with STUN, ICE and TURN with the: - Client implementation of PJSIP (STUN, ICE, TURN) - Server implementation of FS + TURNSERVER.org (that provide STUN/TURN) As per today the PJSIP guys have tested a working setup with: - OpenSER/OpenSIPS - TURNSERVER.org - RTPProxy Does anyone had experience in making the setup with FS + TURNSERVER ? At that point i am thinking that the only choice i have given by default-to-extreme-scenario is not to use anymore the NAT detection and relaying features of FS but roll back to STUN/ICE/TURN solution using FS only for the SIP, but i would like to avoid learning another server product now we are fine with FS. Fabio

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mod_skinny
by Dave Stevenson 29 Mar '10

29 Mar '10

Hi, is mod_skinny (SCCP) still supported on FreeSWITCH ? I see it in the Wiki, but it does not seem to be in any of the modules source directories. regards Dave

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Re: [Freeswitch-users] Set the outgoing caller id
by Peter Olsson 29 Mar '10

29 Mar '10

Most providers also have some kind of screening on caller id numbers. They will only allow your own numbers, and maybe in a specific format. /Peter ________________________________ Från: Erkan Ünlü <erkan(a)speedingtrade.com> Skickat: den 29 mars 2010 19:23 Till: freeswitch-users(a)lists.freeswitch.org <freeswitch-users(a)lists.freeswitch.org> Ämne: Re: [Freeswitch-users] Set the outgoing caller id For internal calls between sipphones that work. <extension name="internal"> <condition field="destination_number" expression="^(\d{4})$"> <action application="set" data="effective_caller_id_name=+49123456789"/> <action application="set" data="effective_caller_id_number=+4912345678"/> <action application="bridge" data="USER/$1@$${domain}"/> </condition> </extension> But for external calls doesn’t work. So set actions are proceed, but not for external calls. I try different and more variables. Kind regards. Erkan From: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of David Ponzone Sent: Monday, March 29, 2010 5:57 PM To: freeswitch-users(a)lists.freeswitch.org Subject: Re: [Freeswitch-users] Set the outgoing caller id Erkan, have you tried to set: <param name="caller-id-in-from" value="true"/> in the gateway params ? Also, your provider is supposed to tell you which field they will trust as the CLID. David Ponzone Direction Technique email: david.ponzone(a)ipeva.fr<mailto:david.ponzone@ipeva.fr> tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr<BLOCKED::http://www.ipeva.fr/> - www.ipeva-studio.com<BLOCKED::http://www.ipeva-studio.com/> Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur. Le 29/03/2010 à 16:40, Erkan Ünlü a écrit : The outgoing caller id don’t work. I try many different fields/variables and also analyze your web link about the Channel Variables. The outgoing id works for internal calls. But for external calls don’t worked. My provider support the outgoing caller id. Here my example from Asterisk. That works. exten => _00XX.,1,Set(CALLERID(num)=00492222222) exten => _00XX.,n,Set(CALLERID(name)=00492222222) …. The variables caller_id_name ande caller_id_number are read only. Can we open it so I can write into the fields, because with effective_caller_id_name and number don’t work it. Origination_caller…… are the same. Hmmm… I don’t understand where is my failure. Kind regards Erkan From: freeswitch-users-bounces(a)lists.freeswitch.org<mailto:freeswitch-users-bounces@lists.freeswitch.org> [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Michael Collins Sent: Friday, March 26, 2010 5:37 PM To: freeswitch-users(a)lists.freeswitch.org<mailto:freeswitch-users@lists.freeswitch.org> Subject: Re: [Freeswitch-users] Set the outgoing caller id Some providers have different ways of handling this. Using "effective_caller_id_number" and name are the usual ways of doing it. However some providers pick up the caller ID from a different field. Try experimenting with this: http://wiki.freeswitch.org/wiki/Channel_Variables#sip_cid_type Perhaps a bridge line like this would work: <action application="bridge" data="{sip_cid_type=rpid}sofia/gateway/voicetrading.com/00000$1<http://voicetrading.com/00000$1>"/> OR <action application="bridge" data="{sip_cid_type=pid}sofia/gateway/voicetrading.com/00000$1<http://voicetrading.com/00000$1>"/> Please try it and let us know what happens. -MC 2010/3/25 Erkan Ünlü <erkan(a)speedingtrade.com<mailto:erkan@speedingtrade.com>> Hi all, I worked since 2 years with asterisk. But now I must use the Windows Platform. So I found Freeswitch. The first works with Freeswitch are very good, but I can’t set the outgoing caller id. The outgoing caller id is shown at the phone on the end of the line. I try some different variables but without result. My voip provider support to set the callerid, in asterisk it works. <action application="bridge" data="sofia/gateway/voicetrading.com/00000$1<http://voicetrading.com/00000$1>"/> …. <include> <context name="default"> <extension name="voicetrading.com<http://voicetrading.com>"> <condition field="destination_number" expression="^00(\d+)$"> <action application="set" data="effective_caller_id_name=009012345678"/> <action application="set" data="effective_caller_id_number=009012345678"/> <action application="set" data="outbound_caller_id_name=009012345678"/> <action application="set" data="outbound_caller_id_number=009012345678"/> <action application="bridge" data="sofia/gateway/voicetrading.com/00000$1<http://voicetrading.com/00000$1>"/> </condition> </extension> …. Can anyone give me a idea for what can I do. Kind regards Erkan Unlu _______________________________________________ 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 _______________________________________________ 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:4bb0e21032931608113207!

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Loopback echo test + conferencing
by Fernando Testa 29 Mar '10

29 Mar '10

Hi I have a running conferencing application and need to measure the total delay (FS+network). I already have a way to measure delay using one FS instance, say FS1, calling an echo application on FS2. This is achieved by recording the session on FS1 and analyzing the cross correlation delay with a octave script. But how do I make FS1 call FS2 where FS2 has conference+echo so conference is included on the test? Thank you! -- Fernando Gregianin Testa Voice Technology Ltda +55 11 35882166

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luasql on CentOS 5.4
by Neil Patel 29 Mar '10

29 Mar '10

I can't get luasql to compile on CentOS 5.4: [root@topaz luasql-2.1.1]# make gcc -fPIC -O2 -Wall -Wmissing-prototypes -Wmissing-declarations -ansi -pedantic -I../compat/src -I/usr/include/mysql -I/usr/local/include -o src/mysql.so -shared src/luasql.o src/ls_mysql.o -L/usr/local/mysql/lib -lmysqlclient -lz /usr/bin/ld: src/luasql.o: relocation R_X86_64_32 against `a local symbol' can not be used when making a shared object; recompile with -fPIC src/luasql.o: could not read symbols: Bad value collect2: ld returned 1 exit status make: *** [src/mysql.so] Error 1 I've tried adding the -fPIC in the make but still no luck. Also, the FS wiki warns agains compiling with the MySQL driver since it leaks memory. So how do I install luasql for MySQL? Couldn't find any RPMs, can anyone here share one? Thanks, Neil

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need legA talks to listen leg B
by Madovsky 29 Mar '10

29 Mar '10

If I don't talk from the legA no sound can be listen. do I need to set a special vars to listen legB as soon as it answers ? I don't use any vad. Thanks

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OneWay Bridge Channels
by Victor Toofic 29 Mar '10

29 Mar '10

Hi! Does anybody know if there is a way to "bridge" an answered channel to a channel that is in "early media" state? The scenario is like this: Channel A Channel B -------------------> FS -------------> (pre_answer, park) (answer) <---<---<---<---<--- Audio <---<---<--- When the channel A arrives I pre_answer and park it, so it is in "early media" state. After that I originate the channel B. When the channel B answers I just want to relay the audio coming from B to A, I don't care if A can't talk to B at that moment. Is there a way to achieve this? Some ideas? Thnks! - Regards, Víctor Toofic

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ooh323 license is now compatible.
by Brian West 29 Mar '10

29 Mar '10

http://www.obj-sys.com/open/ooh323c_kits.php The license of this library changed recently its compatible with ours now. If someone wants to create an endpoint module using this h323 stack then have at it. Just an FYI, /b

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FS auto nat detection systems: are an attempt to provide alternative to STUN/ICE/TURN ?
by Fabio Pietrosanti (naif) 29 Mar '10

29 Mar '10

Hi all, i would like to ask one question to all FS folks. I am really struggling against the automatic nat system of FS, where the full logic is still not really really clear and causing a lot of headheache. Where on asterisk with a simple canreinvite=no everything works fine (even between dirty nat) on FS it does not work and FS start sending RTP to provide ip address on 10.0.0.0/8 (even if there is apply-nat-acl=rfc1918.auto) . However, independently from the bugs of the automatic nat handling systems that have with the time to be noticed, discovered, fixed, etc, etc, my question is: All this method for dynamic nat detection and handling of FS, want to be an alternative to the STUN/ICE/TURN approach of VoIP standard protocol? Fabio

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