June 2011 - FreeSWITCH-users - FreeSWITCH Mailing Lists

SRTP
by Rob Hutton 24 Jun '11

24 Jun '11

I think I have TLS and SRTP working at this point, but in the docs it says to use the following template for the dialplan: http://wiki.freeswitch.org/wiki/Secure_RTP: <extension name="incoming-fxs"> <condition field="destination_number" expression="^(202)$"/> <condition field="${sip_has_crypto}" expression="^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$" break="never"> <action application="set" data="sip_secure_media=true"/> <action application="bridge" data="openzap/1/1"/> </condition> </extension 1) There is a missing > at the end of the close extension tag. 2) There is either a missing / at the end of the internal condition line, or a missing condition close tag somewhere 3) When I fix the interal condition, I get an error: {ERR} mod_dialplan_xml.c:110 Nexted conditions are not allowed All this, but a packet capture shows that SRTP is working based on what I did on: http://wiki.freeswitch.org/wiki/SIP_TLS Can someone give me some guidance on the Secure_RTP page and I will update whatever?

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Recovering from database fault
by Clive Lansink 23 Jun '11

23 Jun '11

Hello list After a computer crash yesterday I noticed FreeSwitch reporting that the database disk image was malformed. After googling on this it seemed that the thing to do was to either delete or rename the db folder and get FreeSwitch to recreate it. After I did that and restarted Freeswitch, I found that all my phones seemed to register ok in that their web interfaces reported they were registered and I could make calls on them, but none of them could ring. When I rang another extension, it seemed from the log that the sofia/internal calling phone would get to the point where it would bridge to the sofia/internal for the extension, and then there would be an error something like could not create the channel, cause, user not registered. So all the phones could make calls but not receive them. I needed to get things working quickly and fortunately I achieved this by brute force - simply uninstalling Freeswitch, getting rid of its folder, installing again, and copying a saved copy of the conf directory back. Our installation is simple and we don't rely on the internal voicemail etc, so that was a quick way to get going again. But does anyone know what was going on here? Are there other data files apart from the db folder we should be aware of? Clive Lansink Email: Clive(a)Lansink.Co.NZ Phone: +64 9 520-4242 Mobile: +64 21 663-999 Fax: +64 21 789-150

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Off Hook Agents in Mod_Fifo
by Zahid Maqbool 23 Jun '11

23 Jun '11

Hi, I've been trying out Freeswitch and so far it seems great, however I wanted to test out call center features. So I had a look at Mod_FIFO. I'd like to understand how to configure off hook agents and if its possible to login an off hook agent via an external application? Looking at the mod FIFO most of the examples are for on hook agents, does anyone know how to configure off hook agents? What exactly I want to do is let's say, there is an agent application on the desktop and I want the agent to put his phone off hook, enter the agent id and extension through this application and click on login. Any pointers would be appreciated. Thanks. Zahid Maqbool

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FreeSWITCHer: issue with ObjectSpace
by Слюсарев Михаил 23 Jun '11

23 Jun '11

Hey, guys My question is about FreeSWITCHer the Ruby framework handle events from FreeSWITCH. I know that this is not exactly what you are discuss here, but I really don't know whith whom to comunicate about this problem. So, if you have experience in FreeSWITCHer and Ruby and would like to help, please read futher. Otherwise, sorry for spam. FreeSWITCHer doesn't sent subscribing messages during of usage of examples/inbound_socket_events.rb. As I can see in the source, that happens because subscribe_to_event function is never trigered. So, the question is why this snippet of code doesn't work as expected? def self.add_event_hook(event, sub_events = [], &block) ObjectSpace.each_object { |e| e.subscribe_to_event(event, sub_events) if e.class.ancestors.include?(FSR::Listener::Inbound) } HOOKS[event] = block end And this is my environment: # cat /etc/redhat-release CentOS release 5.6 (Final) # ruby -v ruby 1.8.5 (2006-08-25) [i386-linux] # gem -v 1.3.5 # gem list *** LOCAL GEMS *** eventmachine (0.12.10) freeswitcher (0.6.12) log4r (1.1.9) Since it's looks like Ruby configuration issue I don't describe other installation steps.

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Re: [Freeswitch-users] FreeTDM - Sangoma B700 - ISDN connection questions - UK
by soapee01 23 Jun '11

23 Jun '11

On 6/6/2011 4:36 PM, John wrote: > Hello, > > > 3. I have built FreeSWITCH from git and installed at /usr/local/... and > then followed the steps on the Ubuntu page in the Wiki to set up the run > control scripts, etc, and run FS non-root as freeswitch:daemon. With > FreeTDM, I have discovered that the /dev/wan* devices are owned by > root:root, and so are inaccessible to FS running as non-root. So for > now I have added a line in /etc/init.d/freeswitch to 'chgrp freeswitch > /dev/wan*'. This is not the most elegant solution, because 'wanrouter > restart' (which seems to be my best friend at the moment) resets the > ownership to root:root. I have tried grepping to see where the mknods > are for these devices, but have been unsuccessful. Is there a better > place to 'permanently' change the device ownership? > John, Have a look at /etc/udev/rules.d You can write a rule there and set the ownership permanently that way. Sorry about the delay on this John. I'd sent it earlier but managed to use the wrong identity and the list didn't bounce my reply (or I didn't notice). You've probably figured this out on your own by now... James

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RTMP/SIP solution - scalability
by Carlo Dimaggio 23 Jun '11

23 Jun '11

Hi all, first of all thank you for the mod_rtmp contribution. It is a very interesting solution that could be used to create UC platform. I would like to know if such a this module is suitable in an hosted environment (with web portal and flash client) and what could be the scalability in this case. I know it is a new/beta module but I'm wondering if somebody have done load test or has thought about this application. Just another question: when we could have documentation about configuration and sip integration? Best Regards, Carlo Dimaggio

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Minor issue with INVITE domain
by David Ponzone 23 Jun '11

23 Jun '11

Hello all, I work with a provider which I need to send calls using their IP (I want to avoid any DNS issues), but I need the FQDN to be in the INVITE domain. So basically, I have a gateway defined with <param name="proxy" value="X.X.X.X"/> but I need the INVITE to be: sip:called@whatever.foo.com and not: sip:called@X.X.X.X I played with everything I could think of: outbound-proxy in gateway params (which seems to do nothing) {sip_invite_domain=whatever.foo.com}bridge/gateway/gw/called also does nothing. I am missing something ? Thanks David Ponzone Direction Technique email: david.ponzone(a)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 - 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.

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Modified Follow me
by Alessandro 23 Jun '11

23 Jun '11

Hi, i'd like to implement this particular behaviour. If I call an extension that isn't registered to FS, FS re-route the call to another destination. If also the second destination in unreachable the call end. I first used the follow me (created with fusionpbx) to implement this feature, but I have the following problem: if the first extension is reachable and the user hangup before answering (hangup in ringing), the follow me procedure starts to contact the second extension in the list. I want that FS start to contact the second phone only if the first extension isn't registered to FS. The second problem is that with follow-me FS answer with 200 OK at the invite of the caller bypassing the ringing status, than the procedure of follow-me is transparent to the caller. I'd like that the caller stay in ringing status until one of the destination extension answer at the call or the call is closed by FS. Any suggestion? Regards Alessandro Luppi -- Ing. Alessandro Luppi Software development Seletech srl Via Collodi 8, 20052 Monza (MI) - Italy Tel: +39.039.5962000 - Fax: +39.039.9716905 email:a.luppi@seletech.com - Web:www.seletech.com orwww.seletech.eu

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Linksys SPA509 MWI Subscribe
by Spencer Thomason 23 Jun '11

23 Jun '11

Hello all, I'm trying to set up a shared VM box with a few Cisco/Linksys SPA509G phones. Does anyone know if it is possible to make these phones subscribe to a mailbox number? I've done it in the past with Polycoms without any issue but I can't seem to get these to work... Thanks for any help! Spencer

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'make current' fails to complete
by elijah 23 Jun '11

23 Jun '11

My attempts to 'make current' against the Git repository are failing today. Can you recommend a method by which I can update my FreeSwitch installation? Here's the the last bit of console logging: ... make[2]: Leaving directory `/usr/src/freeswitch/libs/esl/ruby' make -C java clean make[2]: Entering directory `/usr/src/freeswitch/libs/esl/java' rm -f *.o *.so *~ *.jar make[2]: Leaving directory `/usr/src/freeswitch/libs/esl/java' make -C managed clean make[2]: Entering directory `/usr/src/freeswitch/libs/esl/managed' rm -f *.o *.so *~ make[2]: Leaving directory `/usr/src/freeswitch/libs/esl/managed' make[1]: Leaving directory `/usr/src/freeswitch/libs/esl' make update make[1]: Entering directory `/usr/src/freeswitch' Pulling updates... Updating 5923f71..4bb7683 src/mod/applications/mod_dptools/mod_dptools.c: needs update fatal: Entry 'src/mod/applications/mod_dptools/mod_dptools.c' not uptodate. Cannot merge. make[1]: *** [update] Error 128 make[1]: Leaving directory `/usr/src/freeswitch' make: *** [current] Error 2

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