- FreeSWITCH-users - FreeSWITCH Mailing Lists

Mod_pjsip
by Meftah Tayeb 01 Nov '09

01 Nov '09

Hi my FreeSWITCH friends Today we talked about FreeSWITCH libraries and we found pjsip, which is a lot better than sofia, it can do everything sofia can do and more, it's thread-safe and performs well, I would love to see a mod_pjsip. The library is GPL but they also offer alternative licensing, don't get me wrong, Sofia is great but I think FreeSWITCH will benefit from a thread-safe SIP library. don't say that i need some stuf for free, but we need to make fs a very also we love freeswitch to run in a Embedded PC like the blackfin CPU/boards, Pjsip is very good for Embedded SIP Devicesawesome Open Source project With much love, DelphiWorld____ Information from ESET NOD32 Antivirus, version of virus signature database 4539 (20091024) __________ The message was checked by ESET NOD32 Antivirus. http://www.eset.com __________ Information from ESET NOD32 Antivirus, version of virus signature database 4539 (20091024) __________ The message was checked by ESET NOD32 Antivirus. http://www.eset.com __________ Information from ESET NOD32 Antivirus, version of virus signature database 4539 (20091024) __________ The message was checked by ESET NOD32 Antivirus. http://www.eset.com

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Small bug in switch_ivr_record_file (in trunk)
by Jonas Gauffin 01 Nov '09

01 Nov '09

switch_ivr_play_say.c, line 486. file = switch_core_session_sprintf(session, "%s%s%s%s", switch_str_nil(tfile), tfile ? "]" : "", prefix, SWITCH_PATH_SEPARATOR, file); There should be five %s, not four.

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SIP Proxy with direct media path
by Dmitry Gromov 01 Nov '09

01 Nov '09

Hi! I was a long time user of MySipSwitch service (www.mysipswitch.com) but it looks like they are planning to close their service. I can't use the replacement they have and would actually prefer some locally installed solution. Question - is it possible to use FreeSwitch exactly like MySipSwitch - just SIP Proxy with very flexible dial plan? With MySipSwitch it is possible to: - register service with multiple SIP providers as client - register your ATA with MySipSwitch - create smart dial plan (in Ruby) to route calls based on prefixes and/or call source and some other logic - accept and route calls from all registered providers MySipSwitch does not have any media capabilities - RTP is always established directly between call parties (no re-Invites either). If this is possible - is there any beginner guide for this? Also - I know that FreeSwitch could run in Windows as well as in Linux - would it be possible to use the above functionality in both OSes? Thanks, Dmitry -- DG NJ

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multihomed help
by Joseph L. Casale 01 Nov '09

01 Nov '09

I am setting up fs in pfsense. Following the multihomed tutorial (I also have a dedicated wan/lan int) if I set directory/default.xml domain=10.0.0.1 (my lan int ip) it breaks everything, but if I set <X-PRE-PROCESS cmd="set" data="domain=10.0.0.1"/> in conf/vars.xml I now get audio working correctly, *9999 etc plays MOH. Is there still something I have done wrong? conf/sip_profiles/internal.xml has the int lan ip's entered, and external.xml does not have any edits. Thanks for any info! jlc

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Re: [Freeswitch-users] FreeSWITCH-users Digest, Vol 40, Issue 179
by Ahmed Munir 01 Nov '09

01 Nov '09

Hi, Thanks for reply, it really helped me. One more thing to ask, how can we make decision against >,<, >=, <= in condition header? Like we use == for action and != for anti-action. Kindly highlight it. > > ---------- Forwarded message ---------- > From: Ahmed Munir <ahmedmunir007(a)gmail.com> > To: FreeSwitch <freeswitch-users(a)lists.freeswitch.org> > Date: Wed, 21 Oct 2009 15:37:15 +0500 > Subject: [Freeswitch-users] Call custom variable in condition > Hi, > > I've declared a variable named AUTHENTICATION_STATUS in dialplan, and I > want to use it in condition, as I'm listing down the configuration below; > > <context name="SIP_incoming"> > <extension name="call-sip-extensions"> > <condition field="destination_number" expression="^(\d+)$"> > <action application="set" data="AUTHENTICATION_STATUS=0"/> > <action application="transfer" data="${AUTHENTICATION_STATUS} > XML Authen_Status"/> > </condition> > </extension> > </context> > > <context name="Authen_Status"> > <extension name="exten-auth-status"> > <condition field="AUTHENTICATION_STATUS" expression="^0$"> > <action application="answer"/> > <action application="playback" data="play.wav"/> > </condition> > </extension> > </context> > > > > > But unfortunately it is not working. Kindly advise me how to do implement > it(Note: I don't want to call script). And one more thing to ask how can I > transfer the values within the same context? > > -- > Regards, > > Ahmed Munir > > > > > ---------- Forwarded message ---------- > From: Ghulam Mustafa <mustafa.pk(a)gmail.com> > To: freeswitch-users(a)lists.freeswitch.org > Date: Wed, 21 Oct 2009 15:52:24 +0500 > Subject: Re: [Freeswitch-users] Call custom variable in condition > Ahmed, > > you can't use variables set by "set" application within a condition, though > it doesn't make sense. wondering if there is any logic behind this or it's > just a simple missing feature. anyone? > > -m > > Ahmed Munir wrote: > >> Hi, >> >> I've declared a variable named AUTHENTICATION_STATUS in dialplan, and I >> want to use it in condition, as I'm listing down the configuration below; >> >> <context name="SIP_incoming"> >> <extension name="call-sip-extensions"> >> <condition field="destination_number" expression="^(\d+)$"> >> <action application="set" data="AUTHENTICATION_STATUS=0"/> >> <action application="transfer" data="${AUTHENTICATION_STATUS} >> XML Authen_Status"/> >> </condition> >> </extension> >> </context> >> >> <context name="Authen_Status"> >> <extension name="exten-auth-status"> >> <condition field="AUTHENTICATION_STATUS" expression="^0$"> >> <action application="answer"/> >> <action application="playback" data="play.wav"/> >> </condition> >> </extension> >> </context> >> >> >> >> >> But unfortunately it is not working. Kindly advise me how to do implement >> it(Note: I don't want to call script). And one more thing to ask how can I >> transfer the values within the same context? >> >> -- >> Regards, >> >> Ahmed Munir >> >> >> ------------------------------------------------------------------------ >> >> _______________________________________________ >> 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 >> >> > > > > > > ---------- Forwarded message ---------- > From: Tihomir Culjaga <tculjaga(a)gmail.com> > To: freeswitch-users(a)lists.freeswitch.org > Date: Wed, 21 Oct 2009 13:13:13 +0200 > Subject: Re: [Freeswitch-users] Call custom variable in condition > consider this: > > > > <context name="SIP_incoming"> > <extension name="call-sip-extensions"> > <condition field="destination_number" expression="^(\d+)$"> > <action application="set" data="AUTHENTICATION_STATUS=0"/> > <action application="transfer" data="${AUTHENTICATION_STATUS} > XML Authen_Status"/> > </condition> > </extension> > </context> > > > > <context name="Authen_Status"> > <extension name="exten-auth-status"> > <condition field="${AUTHENTICATION_STATUS}" expression="^0$"> > <action application="answer"/> > <action application="playback" data="play.wav"/> > </condition> > </extension> > </context> > > > > > > here is one of my dialplan. I'm using execute_extension but it is quite the > same... > > > > <extension name="ServiceLookup"> > <condition field="destination_number" expression="(^300030)(.*)"> > <action application="lookup_service_destination" data="in > ${caller_id_number:6:16}, in ${caller_id_number:0:6}, in $2, in $ > 1, in ${network_addr}:5060, out red_contact, out authResult"/> > <action application="log" data="INFO ######################## > ServiceLookup ########################\n"/> > <action application="log" data="INFO ######################## > contact = '${red_contact}' ##############\n"/> > <action application="log" data="INFO ######################## > CallerNum = '${caller_id_number:6:16}' ##########\n"/> > <action application="log" data="INFO ######################## > RADIUS auth = '${authResult}' ##########\n"/> > > <action application="execute_extension" data="doRedirect XML > public"/> > </condition> > </extension> > > > <extension name="doRedirect"> > <condition field="destination_number" expression="^doRedirect$"/> > <condition field="${authResult}" expression="^0$|^60$"> > <action application="log" data="INFO ######################## > RADIUS auth OK!!!' ##########\n"/> > <action application="redirect" data="${red_contact}"/> > <anti-action application="log" data="INFO ######################## > RADIUS auth NOK!! ##########\n"/> > <anti-action application="respond" data="403 Forbidden"/> > </condition> > > </extension> > > > > > On Wed, Oct 21, 2009 at 12:37 PM, Ahmed Munir <ahmedmunir007(a)gmail.com>wrote: > >> Hi, >> >> I've declared a variable named AUTHENTICATION_STATUS in dialplan, and I >> want to use it in condition, as I'm listing down the configuration below; >> >> <context name="SIP_incoming"> >> <extension name="call-sip-extensions"> >> <condition field="destination_number" expression="^(\d+)$"> >> <action application="set" data="AUTHENTICATION_STATUS=0"/> >> <action application="transfer" data="${AUTHENTICATION_STATUS} >> XML Authen_Status"/> >> </condition> >> </extension> >> </context> >> >> <context name="Authen_Status"> >> <extension name="exten-auth-status"> >> <condition field="AUTHENTICATION_STATUS" expression="^0$"> >> <action application="answer"/> >> <action application="playback" data="play.wav"/> >> </condition> >> </extension> >> </context> >> >> >> >> >> But unfortunately it is not working. Kindly advise me how to do implement >> it(Note: I don't want to call script). And one more thing to ask how can I >> transfer the values within the same context? >> >> -- >> Regards, >> >> Ahmed Munir >> >> >> >> _______________________________________________ >> 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 >> >> > > _______________________________________________ > 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 > > -- Regards, Ahmed Munir

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mod_t38gateway
by Tihomir Culjaga 01 Nov '09

01 Nov '09

does anybody know how does it work and how to use it in a dialplan? freeswitch@nemesis> freeswitch@nemesis> freeswitch@nemesis> load mod_t38gateway API CALL [load(mod_t38gateway)] output: +OK 2009-10-30 22:44:38.204268 [NOTICE] mod_t38gateway.c:147 T.38 gateway enabled 2009-10-30 22:44:38.204268 [CONSOLE] switch_loadable_module.c:889 Successfully Loaded [mod_t38gateway] 2009-10-30 22:44:38.204268 [NOTICE] switch_loadable_module.c:248 Adding Application 't38gateway' 2009-10-30 22:44:38.205374 [NOTICE] switch_loadable_module.c:270 Adding API Function 't38gateway' freeswitch@nemesis> freeswitch@nemesis> freeswitch@nemesis> freeswitch@nemesis> t38gateway API CALL [t38gateway()] output: -USAGE: <uuid> <command> freeswitch@nemesis> T.

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ANNOUNCEMENT: FreeSWITCH 1.0.5pre3 is available for testing!
by Michael Collins 01 Nov '09

01 Nov '09

Hello FreeSWITCHers, The latest FreeSWITCH version is now available for download on the files<http://files.freeswitch.org/freeswitch-1.0.5pre3.tar.gz>site. The announcement story is in the main FreeSWITCH <http://www.freeswitch.org/node/212> site. Please download and test, and then test some more. We need your feedback. The sooner we get your feedback, the more quickly we can roll the official 1.0.5. Thanks! -Michael

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[mod_socket] Can't set channel variable SOMETIMES
by Artem Shiyanov 31 Oct '09

31 Oct '09

Hello! I've got strange problem: In my app which talks to FreeSwitch via mod_socket there is such logic: <pre conditions: channel1 - newly created, parked channell - with help of 'uuid_create' and 'originate ... &park'; channel2 is talking with channel3> for (channel in {array of channel1, channel2, channel3}) { set value for the channel variable X } uuid_transfer channel1 extension Y uuid_transfer -both channel2 extension Y In extension Y there is a logic based on the channel variable X. The problem itself: every second launching falls, the reason - channel variable X hasn't been set for one of channel1..3. Sniffer shows that commands are sent (and confirmated by FreeSwitch) consiquently, FreeSwitch sends OK for each command in time. Temporary sollution that I found is to wait 0.5 sec after 'set' commands have been sent and before transfer any channel to extension Y. Maybe some already faced with similar problem? Thanks, Artem

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New FreeSwitch User - WAPSERVER Question
by Dave Stevenson 31 Oct '09

31 Oct '09

Hi, I am new to FreeSwitch - and VOIP too (as the following questions will indicate). I am trying to setup a VOIP system at home and am just getting started, I have been trawling through the Docs/Wiki and realise that I have a lot to learn ! Anyway, here's the first (of what I'm sure will be many !) questions, hopefully someone will take the type to help me out...... OK, I have already worked out that FreeSwitch seems to be much better supported under Linux than Windows (is that a fair comment ?), but while I have a long term intention to give Linux a go, it's something that is on the back burner (and has been for some time). At the moment, the current plan is to use FreeSwitch under WindowsXP and I have loaded the latest Windows Install. Everything seems to be working fine so far - well, in test mode anyway, my VOIP phones and teleco interface won't be here until next week, so at the moment, I'm just trying out the system with X-Lite. The first question that I have is about WAPSERVER. This was installed as part of the FreeSwitch install. At the moment, I am manually starting FreeSwitch and WAPSERVER does not seem to start automatically. When I manually start it, it says that it is OFFLINE until I manually put it ONLINE. So :- 1. What does WAPSERVER actually do ? 2. How should it be started ? 3. My next step is to configure the Server PC to auto-logon to my Windows 2003 Server domain and put FreeSwitch in the Startup Group - do I need to add WAPSERVER there too ? The next question is about the ServerIP address. Not thinking far enough ahead, the machine that I installed FreeSwitch on uses DHCP to get its IP Address. I guess that the FreeSwitch server should really have a static IP Address. If I change the PC to have a static IP Address, then 1. Do I need to re-install FreeSwitch 2. If not, is there a or is there command to get FreeSwitch to updated itself with a new IP Address ? 3. Maybe the Server does not need to know the change - maybe if only affect the clients (Phones/SoftPhones ?) Thanks & Best Regards Dave

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no REINVITE on Blind Transfer with bypass_media
by Humberto Quintana 30 Oct '09

30 Oct '09

Hi everybody, I'd like some help with this situation that is 'haunting' me :-) My scenario is as follows: inbound-bypass-media is set in the profile because we dont want FS handling the media. 1. A calls B 2. FS sends to B the A's SDP 3. B answers 4. FS sends to A the B's SDP 5. Media going directly between A and B 6. B REFERs the call to C (blind transfer with no reINVITE for Hold) 7. FS accepts(202) the REFER and sends the NOTIFY 7a. B and FS send the BYE 8. FS sends an INVITE to C with A's SDP 9. C answers 10. FS doesn't send a reINVITE to A to let it know about C's SDP Is that the expected FS behavior or is this a bug? I'd appreciate any hints. Thanks, Humberto _________________________________________________________________ Lots of fantastic Windows 7 offers, in one convenient place. Get the perfect deal for you now. http://go.microsoft.com/?linkid=9691633

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