- FreeSWITCH-users - FreeSWITCH Mailing Lists

TLS client authentification
by Alexandre Savard 19 Oct '09

19 Oct '09

Hi, Does Freeswitch support TLS Client-Authenticated handshake. Openssl does, but it has to be enabled in order to send the certificate request to the client. I tested simple TLS hanshaking and it wotks well. Thanks.

2 1

sofia gateways and linux multipath routing
by François Delawarde 19 Oct '09

19 Oct '09

Hello all, I'm interested in using mod_sofia with multiple Internet connections (configured as a unique load-balancing route using multipath). One solution would be to define a different profile for each connection, but it would be more practical having a unique external profile that would automatically handle everything (detecting multiple public IPs and selecting the right one for a call, being able to select the router for gateway registration...). Routing table: 192.168.10.0/24 dev eth0 proto kernel scope link src 192.168.10.1 192.168.1.0/24 dev eth1 proto kernel scope link src 192.168.1.2 192.168.2.0/24 dev eth2 proto kernel scope link src 192.168.2.2 default proto static nexthop via 192.168.1.1 dev eth1 weight 1 nexthop via 192.168.2.1 dev eth2 weight 1 Both default routers (192.168.1.1 and 192.168.2.1) would have a distinct public IP. Several questions cross my mind: - Can a unique sofia profile be bound to multiple IPs (not 0.0.0.0)? - How would FS behave with a unique external profile in that situation? * Would FS reply to an incoming call using the same router it came from forcing packet source address? * Would FS stick to a unique router for all flows of an outgoing call (SIP, RTP, UDPTL)? * Can I force a gateway to use a given router (for calls, registration, ...)? * Would the NAT system (using stun or auto-nat) work in that situation, or does it assume only one default router (and a unique public IP) exists per profile? - Knowing the above, would it be necessary to use a different profile for each router/interface, and define the same gateway in each of these? - Tricky question: What if multiple routers are on the same network/interface (192.168.1.1, 192.168.1.2, ...)? Thanks in advance, François.

2 1

Freeswitch.managed
by srinivasula reddy 19 Oct '09

19 Oct '09

Hi, How can i use freeswitch.managed project. what are the parameters for calling Execute method? and how can i call? any help -- Srinivasula Reddy K

2 2

Re: [Freeswitch-users] NAT problems migrating from Version 1.0.trunk (13168M) to Version 1.0.trunk (15166)
by Chris Fowler 19 Oct '09

19 Oct '09

I've tried all sorts of debug and parameter changes over the weekend, but still can't figure out the correct solution. If I disable timers on the sip profile then all works fine. <param name="enable-timer" value="false"/> But that seems like a hack; not a correct solution. With the build 13168M (which is pre the new NAT functionality) everything worked fine. The SIP trace shows the phones and FreeSWITCH happily exchanging NOTIFY and "200 OK" messages. Audio's working - just calls timeout after 100 seconds with RECOVERY_ON_TIMER_EXPIRE. Is enforcement of this timer new functionality - and really just exposing a problem I've always had before? The config is (50 Polycom Phones -> NAT -> Internet -> Amazon EC2) I would really appreciate some pointers on what to look for; additional trace that might reveal something. Thanks, Chris. On Fri, 16 Oct 2009 20:06:52 -0700, "Chris Fowler" <chris(a)fowler.cc> said: > Hi, > > We've been using 13168M in production for some time now (works great). > I want to get us onto the latest build but am having problems getting > NAT to work. > > Phones can register; can dial 9999 test #, but after 100 seconds the > call is disconnected with error: > 2009-10-16 19:52:26.936618 [NOTICE] sofia.c:4038 Hangup > sofia/internal/1001(a)myhost.mydomain.com [CS_EXECUTE] > [RECOVERY_ON_TIMER_EXPIRE] > > I took the standard internal.xml and vars.xml files from the new build > and made the following modifications - which worked previously: > > modify conf/vars.xml and update > <X-PRE-PROCESS cmd="set" data="domain=myhost.mydomain.com"/> > > <X-PRE-PROCESS cmd="set" data="bind_server_ip=1.2.3.4"/> > <X-PRE-PROCESS cmd="set" data="external_rtp_ip=1.2.3.4"/> > <X-PRE-PROCESS cmd="set" data="external_sip_ip=1.2.3.4"/> > > Modify conf/sip_profiles/internal.xml > <param name="aggressive-nat-detection" value="true"/> <param > name="ext-rtp-ip" value="$${external_rtp_ip}"/> <param > name="ext-sip-ip" value="$${external_sip_ip}"/> <param > name="NDLB-received-in-nat-reg-contact" value="true"/> <param > name="NDLB-force-rport" value="true"/> <param > name="NDLB-broken-auth-hash" value="true"/> > > The big difference I note is that on PRODUCTION (which works) "sofia > status profile internal" yields: > URL sip:mod_sofia@1.2.3.4:5060 > BIND-URL sip:mod_sofia@1.2.3.4:5060;maddr=10.250.35.224 > > But on Test I see: > URL sip:mod_sofia@10.250.66.210:5060 > BIND-URL sip:mod_sofia@10.250.66.210:5060 > > Any ideas? > > Thanks, Chris. > > > _______________________________________________ > 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

2 1

set up sound on windows
by Timur Irmatov 19 Oct '09

19 Oct '09

Hi! We are currently developing small call-center for our company using FreeSWITCH. Our demands are pretty low at the moment, and we have implemented all necessary features, now moving to testing.. We manage FreeSWITCH through mod_erlang_event. FreeSWITCH and mod_erlang_event are great, I want to thank all developers for those two great pieces of technology! We met strange problem with windows clients though. It is not related to FreeSWITCH, but I don't know where to ask. Tried searching google, but found nothing. As soon as we plug headset with microphone into windows machine, we immediately start hearing environment though headset. It seems as sound from microphone is transferred to earbuds, which is pretty annoying. Does anyone have any hints regarding configuration of audio/mics on Windows? -- Timur Irmatov, xmpp:irmatov@jabber.ru

1 0

Freeswitch 1.0.4 problems with music on hold
by Durk de Beer 19 Oct '09

19 Oct '09

Hello, This is something I came across on Freeswitch 1.0.4 First let me explain what I'm trying to do. I want Free-Switch to behave as a proxy so in the settings section of Sofia.conf.xml I use and As fare as I am able to follow the RTP stream it is passing by Freeswitch making it a proxy. So far so good. Now I want to serv music on hold if a call is put on hold. The only way I am able to find for Freeswitch to serv music on hold (MOH) is for it to be in media. Ok we've got a command for this situation namely Ok if I put the last parram command into the Sofia.conf.xml Freeswitch is starting to behave funny. Let me try to explain this by an example. Let suppose Alice is calling Bob without the last command, the “resume media on hold”. If Bob puts Alice on hold al is silent on Bob's and Alice's phone. If Bob takes Alice of hold there's two way audio. Ok no I put the command in sofia.onf.xml, the “resume media on hold”. Now Alice is calling Bob again and Bob is putting her on hold again. You are expecting MOH on this moment but it isn't, all there is are the sounds of silence. Ok mistake made checking local streams, and they are there?. Yes it is. Ok other mistake, didn't put the “hold-music”command in sofia.conf.xml. Noop it's in there so there must be MOH. Codecs then?. Noop there in vars.xml. Ok for me it is time to throw in the towel, I'll find it out later. Ok Bob is now taking Alice of hold and surprise surprise there is only one way audio, between Alice and Bob. I didn't have the time to look with Wireshark where al the steams go so sorry for that. Did anyone came across the same problem with Freeswitch 1.0.4. I've a 1.0.2 running and that is doing everything a ok. Kind regards, Durk

1 0

freeswitch french community
by Meftah Tayeb 19 Oct '09

19 Oct '09

hello folk we are starting a new freeswitch community for french users the newly created Irc channel is #freeswitch-fr freeswitch staf, please can we register this channel officialy to the freeswitch freenode group? also can we build a french documentation and host it somehere? in the wiki? my long time friend, diegoviola i going to help about the freeswitch wiki markup we are waiting for french users to join, and welcome thank toAnthony Minessale for accepting the community welcome! __________ Information from ESET NOD32 Antivirus, version of virus signature database 4520 (20091018) __________ The message was checked by ESET NOD32 Antivirus. http://www.eset.com

4 5

Outbound sip address call fails
by Lars Zeb 19 Oct '09

19 Oct '09

I am trying to make a test call to the FreeSwitch conference address, sip:888@conference.freeswitch.org on a Bria softphone. However, it fails with a 404 Not found message. I have struggled with this for a while. Is there something special I must do on the Bria to make this happen correctly? Thanks, Lars http://pastebin.freeswitch.org/10715

4 12

macros for performing repeated dialplan actions?
by Simon J Mudd 18 Oct '09

18 Oct '09

Hi again, I'm trying to build up a soho dialplan to replicate an existing Asterisk setup I want to replace. I'm editing /usr/local/freeswitch/conf/dialplan/default/00_mydomain.com.xml to add some dialplan rules for various destination numbers I'm using. These dialplans may use one of several gateways I have configured. I currently have: <include> <extension name="spanish_mobiles.mydomain.com"> <condition field="destination_number" expression="^(6\d{8})$"> <action application="set" data="effective_caller_id_number=${outbound_caller_id_number}"/> <action application="set" data="effective_caller_id_name=${outbound_caller_id_name}"/> <action application="bridge" data="sofia/gateway/asterisk/$1"/> </condition> </extension> <extension name="spanish_fijos.mydomain.com"> <condition field="destination_number" expression="^(9\d{8})$"> <action application="set" data="effective_caller_id_number=${outbound_caller_id_number}"/> <action application="set" data="effective_caller_id_name=${outbound_caller_id_name}"/> <action application="bridge" data="sofia/gateway/asterisk/$1"/> </condition> </extension> ... more stuff not shown ... </include> Which routes out to my existing asterisk installation. I now want to record all calls. Looking at http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_record_session it seems I need to add a section like into each entry prior to the application="bridge". <action application="set" data="gateway_name=XXXXX"/> <action application="set" data="RECORD_TITLE=Recording ${destination_number} ${caller_id_number} ${strftime(%Y-%m-%d %H:%M)}"/> <action application="set" data="RECORD_COPYRIGHT=(C) 2009 my data here."/> <action application="set" data="RECORD_SOFTWARE=FreeSwitch"/> <action application="set" data="RECORD_ARTIST=My name"/> <action application="set" data="RECORD_COMMENT=My comment"/> <action application="set" data="RECORD_DATE=${strftime(%Y-%m-%d %H:%M)}"/> <action application="set" data="RECORD_STEREO=true"/> <action application="record_session" data="$${base_dir}/recordings/${gateway_name}_${strftime(%Y-%m-%d-%H-%M-%S)}_${destination_number}_${caller_id_number}.wav"/> My question is how do I avoid repeatedly using these same 8-9 lines and "merge" this into some macro which given the right parameters allows me to define various extensions using different gateways but all configured "the same way"? Thanks for any pointers to how this should be done. Simon

4 6

anyway to not to retry
by Juan Backson 18 Oct '09

18 Oct '09

Hi, Is there anyway to configure freeswitch so that it won't retry the SIP message when 200 OK is not received? thanks, jb

2 2