- FreeSWITCH-users - FreeSWITCH Mailing Lists

Free switch + Google Voice + Cisco Call Manager
by Robbie A. Garrett 17 Mar '12

17 Mar '12

Hello Everyone, I have noticed a lack of documentation when it comes to cisco call manager and free switch integration. I would be willing to assist with documentation for the WiKI... I need to be point in the right direction though. I would like to create some documentation that would assist with the following 1) Google voice as the "carrier", Free switch as the "gateway", and call manager as the switch. 2) Call manager as the switch and free switch as the "gateway"

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Re: [Freeswitch-users] FreeSWITCH-users Digest, Vol 69, Issue 127
by virendra bhati 17 Mar '12

17 Mar '12

Hi Patrick , I tryed a lot effort on google but no help i got and even i sent e-mail to lua mail list as well. but no one replied on it. Sorry but if you reply to me with details on luac then i will be a big help for me... On Sat, Mar 17, 2012 at 2:10 AM, < freeswitch-users-request(a)lists.freeswitch.org> wrote: > Send FreeSWITCH-users mailing list submissions to > freeswitch-users(a)lists.freeswitch.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > or, via email, send a message with subject or body 'help' to > freeswitch-users-request(a)lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner(a)lists.freeswitch.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of FreeSWITCH-users digest..." > > Today's Topics: > > 1. Re: blocking destination number (Brian Foster) > 2. Re: Mod_flite wont load (Jeff Lenk) > 3. Re: Any help me to installed Luac compiler (Patrick Lists) > 4. mod_shout issue (Balamurugan Mahendran) > 5. Strange VLAN problem with freeswitch (Erin O'Meara) > 6. Re: Getting real-time amplitude of a channel (Michael Collins) > > > ---------- Forwarded message ---------- > From: Brian Foster <bdfoster(a)endigotech.com> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Fri, 16 Mar 2012 12:58:27 -0400 > Subject: Re: [Freeswitch-users] blocking destination number > > The toll_allow variable.would be defined in the actual user xml file > located in /usr/local/freeswitch/conf/directory. Then you would evaluate > that variable in the dialplan that would end up making the bridge to your > gateway. > > -BDF > On Mar 16, 2012 5:21 AM, "Miha" <miha(a)softnet.si> wrote: > >> Hi @Gabriel, >> >> I was looking on wiki how to define new xml or document where I will >> have my restricted numbers. >> Where are domestic,international,local defined? >> >> Thanks! >> Miha >> >> On 3/10/2012 11:23 PM, Gabriel Gunderson wrote: >> > On Fri, Mar 9, 2012 at 7:37 AM, Miha Zoubek<miha(a)softnet.si> wrote: >> >> what is the bast way to block destination number for certain user. >> >> Is it possible to do it in user/dir? >> > Ultimately, it would be a dialplan configuration, but you could set >> > the variable that you match in the directory. See the example of >> > 'toll_allow' in the default FreeSWITCH configuration... >> > >> > >> > This is set on a per-user basis in the DIRECTORY: >> > >> > <variable name="toll_allow" value="domestic,international,local"/> >> > >> > >> > >> > But it's considered while evaluating the DIALPLAN: >> > >> > <extension name="local.example.com"> >> > <condition field="${toll_allow}" expression="local"/> >> > <condition field="destination_number" expression="^(\d{7})$"> >> > <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/${default_gateway}/1${default_areacode}$1"/> >> > </condition> >> > </extension> >> > >> > >> > >> > You could easily do something similar by having a list of numbers that >> > they can't call listed in the directory (maybe something like >> > 'restricted_numbers') and check to make sure the destination doesn't >> > match it in the dialplan. >> > >> > Good luck! >> > >> > >> > Gabe >> > >> > >> _________________________________________________________________________ >> > Professional FreeSWITCH Consulting Services: >> > consulting(a)freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > FreeSWITCH-powered IP PBX: The CudaTel Communication Server >> > http://www.cudatel.com >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://wiki.freeswitch.org >> > http://www.cluecon.com >> > >> > 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 >> > >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting(a)freeswitch.org >> http://www.freeswitchsolutions.com >> >> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >> http://www.cudatel.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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: Jeff Lenk <jeff(a)jefflenk.com> > To: freeswitch-users(a)lists.freeswitch.org > Cc: > Date: Fri, 16 Mar 2012 10:46:10 -0700 (PDT) > Subject: Re: [Freeswitch-users] Mod_flite wont load > Please try git head and let me know if that works. > > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/Mod-flite-wont-load-tp7376926… > Sent from the freeswitch-users mailing list archive at Nabble.com. > > > > > ---------- Forwarded message ---------- > From: Patrick Lists <freeswitch-list(a)puzzled.xs4all.nl> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Fri, 16 Mar 2012 18:54:19 +0100 > Subject: Re: [Freeswitch-users] Any help me to installed Luac compiler > On 16-03-12 06:07, virendra bhati wrote: > >> Hi list, >> >> I don't find any link on which luac compiler is mention. only one link >> was found on which description about Luac was mention. >> http://www.lua.org/manual/4.0/**luac.html<http://www.lua.org/manual/4.0/luac.html> >> >> Anyone know how to installed Luac compiler in Centos ? >> > > Can't you try a little harder to figure out these basic questions > yourself? Did you even try to Google for "luac CentOS"? That would have > given you a hint. > > Since your question is not a FreeSWITCH question but it is a lua question > why not ask on the lua mailinglist or on the lua irc channel if they have > one. > > Regards, > Patrick > > > > > ---------- Forwarded message ---------- > From: Balamurugan Mahendran <b2m(a)a-cti.com> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Fri, 16 Mar 2012 23:34:23 +0530 > Subject: [Freeswitch-users] mod_shout issue > Hi all, > > 2012-03-16 18:02:32.957534 [ERR] mod_shout.c:859 Error: MPG123 Error at > /usr/src/freeswitch/src/mod/formats/mod_shout/mod_shout.c:659. > 2012-03-16 18:02:32.957534 [ERR] mod_shout.c:862 Error from mpg123: > Invalid mpg123 handle. (code 10) > > Please let me know what Iam missing here. > > Thanks, > Bala > > > ---------- Forwarded message ---------- > From: "Erin O'Meara" <erin.omeara(a)salmonbaytechnology.com> > To: freeswitch-users(a)lists.freeswitch.org > Cc: > Date: Fri, 16 Mar 2012 09:58:51 -0700 > Subject: [Freeswitch-users] Strange VLAN problem with freeswitch > I have moved offices down the street and extended my office network via > vlan (old office was using VLAN also). I have two networks one internal and > one wireless that I sell wireless service with. > > FusionPBX (In the cloud) -> Cable Modem -> PFSense -> VLAN -> Phone. > > my phone works fine for a few days on either network then calls stop > working, if I switch the phone VLAN to the other VLAN it starts working > again. I noticed this issue from my office that is connect of a Ubiquiti > wireless link and VLAN extended to my new office. > > What happens is the calls ring and I can pickup but no audio in either > direction. > > here is the pastebin of a call failing then I plugged the same phone into > the other vlan and reboot and made a successful call. > > http://pastebin.freeswitch.org/18675 > > Regards, > > 206.905.9520 > http://salmonbaytechnology.com > > > > ---------- Forwarded message ---------- > From: Michael Collins <msc(a)freeswitch.org> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Fri, 16 Mar 2012 13:39:55 -0700 > Subject: Re: [Freeswitch-users] Getting real-time amplitude of a channel > Post the bounty on the wiki and then shoot an email to the -dev list > letting them know there's a new bounty available... > > -MC > > On Fri, Mar 16, 2012 at 9:59 AM, Matt Stockton <mstockton(a)harqen.com>wrote: > >> Hi Michael, >> >> Thanks for the info!! Is there information I can find on how the bounties >> work? I'm interesting in getting this feature, but it would probably take >> me much longer to implement than someone who knows the code well. I see a >> bounty wiki entry...is that active? or is there a better place to post such >> things? >> >> Thanks, >> Matt >> >> >> On Wed, Mar 14, 2012 at 12:30 PM, Michael Collins <msc(a)freeswitch.org>wrote: >> >>> AFAIK there isn't anything pre-rolled that you can use. However, I know >>> that Tony's auto gain control (agc) logic checks the "volume" level coming >>> in on a channel. You might be able to look in mod_conference.c and see >>> where the agc calculates the incoming volume level and write an API that >>> displays that value. >>> >>> I'd start by looking at the function check_agc_levels() and for >>> references to member->agc_volume_in_level. Hopefully that will give you a >>> place to start. >>> >>> -MC >>> >>> On Wed, Mar 14, 2012 at 10:26 AM, Matt Stockton <mstockton(a)harqen.com>wrote: >>> >>>> Hi all, >>>> >>>> I am interested in providing a visualization of a channel in a >>>> conference. I am able to process the start talking and stop talking events >>>> from the conference, but I'm wondering, has anyone done anything that can >>>> get the real-time amplitude of a channel? I didn't see any built-in API >>>> commands to do this, but wondering if there's any modules or something else >>>> that I'm overlooking? Or maybe now is the time for me to dig into the API >>>> documentation? >>>> >>>> Just curious if anyone has done something like this before, and if so, >>>> if you could guide me in the right direction. >>>> >>>> Thanks! >>>> Matt >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting(a)freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>> http://www.cudatel.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> 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 >>>> >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting(a)freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>> http://www.cudatel.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> 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 >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting(a)freeswitch.org >> http://www.freeswitchsolutions.com >> >> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >> http://www.cudatel.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 > > -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer E-mail-: virbhati(a)gmail.com Skype id:- virbhati2 Hyderabad(India)

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Need Help to handle message
by Balamurugan Mahendran 17 Mar '12

17 Mar '12

where can I find more details about nua? tport_wakeup_pri(0xa7c160): events IN tport_recv_event(0xa7c160) tport_recv_iovec(0xa7c160) msg 0x7f18080037b0 from (udp/10.248.13.3:5060) has 487 bytes, veclen = 1 tport_deliver(0xa7c160): msg 0x7f18080037b0 (487 bytes) from udp/ 23.21.230.74:5060/sip next=(nil) nta: received MESSAGE sip:+15032721367@50.12.112.39 SIP/2.0 (CSeq 102) nta: Via check: received=23.21.230.74 nta: canonizing sip:+15032721367@50.12.112.39 with contact nta: MESSAGE (102) going to a default leg nua: nua_stack_process_request: entering nua: nh_create: entering nua: nh_create_handle: entering nua: nua_stack_set_params: entering soa_clone(static::0xa69c50, 0xa69490, 0x7f17f80f5180) called soa_set_params(static::0x7f17f80f52a0, ...) called nta_leg_tcreate(0x7f17f8088d30) tport_tsend(0xa7c160) tpn = UDP/23.211.230.74:5060 tport_resolve addrinfo = 23.211.230.74:5060 tport_by_addrinfo(0xa7c160): not found by name UDP/23.211.230.74:5060 tport_vsend(0xa7c160): 539 bytes of 539 to udp/23.211.230.74:5060 tport_vsend returned 539 nta: sent 200 OK for MESSAGE (102) nua(0x7f17f80f5180): event i_message 200 OK nua: nua_application_event: entering nua: nua_handle_magic: entering nua: nua_respond: entering nua(0x7f17f80f5180): sent signal r_respond nua: nua_handle_magic: entering nua: nua_handle_destroy: entering nua(0x7f17f80f5180): sent signal r_destroy nua(0x7f17f80f5180): recv signal r_respond 202 Accepted nua(0x7f17f80f5180): event i_error 500 Responding to a Non-Existing Request nua(0x7f17f80f5180): recv signal r_destroy nta_leg_destroy(0x7f17f8088d30) soa_destroy(static::0x7f17f80f52a0) called nta: timer J fired, terminate 200 response incoming_reclaim_all((nil), (nil), 0x7f180752ec40) nta_incoming_timer: 0/0 resent, 0/0 tout, 1/2 term, 1/2 free nta: timer set next to 14002 ms tport_wakeup_pri(0xa7c160): events IN tport_recv_event(0xa7c160) tport_recv_iovec(0xa7c160) msg 0x7f1808147e10 from (udp/10.248.13.3:5060) has 4 bytes, veclen = 1 tport_deliver(0xa7c160): bad msg 0x7f1808147e10 (4 bytes) from udp/ 115.11.5.34:5060/sip next=(nil) nta: timer J fired, terminate 200 response incoming_reclaim_all((nil), (nil), 0x7f180752ec40) nta_incoming_timer: 0/0 resent, 0/0 tout, 1/1 term, 1/1 free nta: timer not set tport_wakeup_pri(0xa7c160): events IN tport_recv_event(0xa7c160) tport_recv_iovec(0xa7c160) msg 0x7f1808147e10 from (udp/10.248.13.3:5060) has 4 bytes, veclen = 1 tport_deliver(0xa7c160): bad msg 0x7f1808147e10 (4 bytes) from udp/ 115.11.5.34:5060/sip next=(nil) freeswitch@internal> Thanks, Bala

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Send calls through registered gateway
by David Villasmil 17 Mar '12

17 Mar '12

Hello guys, I seem to be unable to send calls through a registered gateway... is there any sample dialplan available? thanks David

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Getting real-time amplitude of a channel
by Matt Stockton 16 Mar '12

16 Mar '12

Hi all, I am interested in providing a visualization of a channel in a conference. I am able to process the start talking and stop talking events from the conference, but I'm wondering, has anyone done anything that can get the real-time amplitude of a channel? I didn't see any built-in API commands to do this, but wondering if there's any modules or something else that I'm overlooking? Or maybe now is the time for me to dig into the API documentation? Just curious if anyone has done something like this before, and if so, if you could guide me in the right direction. Thanks! Matt

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mod_shout issue
by Balamurugan Mahendran 16 Mar '12

16 Mar '12

Hi all, 2012-03-16 18:02:32.957534 [ERR] mod_shout.c:859 Error: MPG123 Error at /usr/src/freeswitch/src/mod/formats/mod_shout/mod_shout.c:659. 2012-03-16 18:02:32.957534 [ERR] mod_shout.c:862 Error from mpg123: Invalid mpg123 handle. (code 10) Please let me know what Iam missing here. Thanks, Bala

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Any help me to installed Luac compiler
by virendra bhati 16 Mar '12

16 Mar '12

Hi list, I don't find any link on which luac compiler is mention. only one link was found on which description about Luac was mention. http://www.lua.org/manual/4.0/luac.html Anyone know how to installed Luac compiler in Centos ? -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer E-mail-: virbhati(a)gmail.com Skype id:- virbhati2 Hyderabad(India)

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Strange VLAN problem with freeswitch
by Erin O'Meara 16 Mar '12

16 Mar '12

I have moved offices down the street and extended my office network via vlan (old office was using VLAN also). I have two networks one internal and one wireless that I sell wireless service with. FusionPBX (In the cloud) -> Cable Modem -> PFSense -> VLAN -> Phone. my phone works fine for a few days on either network then calls stop working, if I switch the phone VLAN to the other VLAN it starts working again. I noticed this issue from my office that is connect of a Ubiquiti wireless link and VLAN extended to my new office. What happens is the calls ring and I can pickup but no audio in either direction. here is the pastebin of a call failing then I plugged the same phone into the other vlan and reboot and made a successful call. http://pastebin.freeswitch.org/18675 Regards, 206.905.9520 http://salmonbaytechnology.com

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Record Session
by Joe Flemmings 16 Mar '12

16 Mar '12

How can i record a call in two seperate directories at the same time the following does not work <action application="record_session" data="$${base_dir}/recordings/admin/${strftime(%Y-%m-%d-%H-%M-%S)}_${destination_number}_${caller_id_number}.wav"/> <action application="record_session" data="$${base_dir}/recordings/users/${strftime(%Y-%m-%d-%H-%M-%S)}_${destination_number}_${caller_id_number}.wav"/> Joe

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sips, tls, srtp, etc
by Bzzz 16 Mar '12

16 Mar '12

FS latest FusionPBX ===================== Hi list, I read a lot from the wiki, but I'm a bit lost about securing calls. From what I read elsewhere, ssl+srtp seems to be the best solution, however the wiki only talks about tls+srtp. I modified conf/vars.xml as of the wiki & enabled xxx_ssl_enable, but put 'sslv23' instead of 'tls'; I also modified conf/directory/default.xml (from 'tls' to 'sslv23') as: <param name="dial-string" value="{sip_secure_media=${regex(${sofia_contact(${dialed_user}@${dialed_domain})}|transport=sslv23)},presence_id=${dialed_user}@${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}" /> is this right & sufficient? If not, what do I miss or what would be the best solution to make sure all calls will be secured both signaling & conversations? At this time, I make tests with the jitsi softphone (ex sip-commu‭nicator) which seems to use SRTP once the SAS has been accepted from each side: that's what wireshark shows, but it also show that SIP is use instead of SIPS, is it because I move 'tls' to 'sslv23'? JY --

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