- FreeSWITCH-users - FreeSWITCH Mailing Lists

Minimum Billing using Nibblebill
by Shahzad Bhatti 22 Apr '14

22 Apr '14

hi everyone, i am trying to impalement increment billing using Mod_nibblebill as : like for mexico rate is 60/60 means minimum duration is 60 sec and next increment duration is also 60 sec or like for Austria Rate is 60/30 means minimum duration is 60 sec and next increment duration is 30 so if call to mexico duration is 15 sec then it cost for 60 sec and id duration is 61 sec then it cost for 120 sec. but in Austria case if call duration is 15 sec then cost it for 60 sec and if duration is 61 sec then it cost 90 sec so while reading wiki i found that i can use <variable name="nibble_increment" value="30" /> for second problem but want to know how i can set for the 1st case that if call duration is 15 second then call cost for 60 seconds. i read here https://wiki.freeswitch.org/wiki/Mod_nibblebill#Billing_a_Call as Billing of minimums does not exist (yet) so i want to know how that i can implement that or tell me how anyone manage that issue. Regards Shahzad Bhatti

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Profiles and Gateways
by Oscar Franzen 22 Apr '14

22 Apr '14

Hi! I am trying to understand the relationship between profiles and gateways. Can someone explain the relationship or direct me in a direction where I can find more info? (I have already tried the wiki and googled quite a bit). When would I want more than on external and one internal profile? Would I ever want less than those two? Do I always want to separate internal and external? When would I add more than one gateway for a profile? I have a SIP trunk and my provider asked me to forward port 5060-5065. So I did that and then I set external_sip_port to 5060. I receive incoming calls on port 5060. Is there a standard way of registering an other port with my SIP trunk provider? As I only have one gateway (a SIP trunk) for my external profile I don't really understand the relationship between the profile and gateway. I don't really understand when I would want more than one gateway for that profile. I also want to use the internal profile for users to register their softphones directly to my freeswitch server. Do I have to do anything special to get this to work? Will there be any problems from letting two profiles listen on the same port (5060)? Regards Oscar Franzén

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Re: [Freeswitch-users] DID providers - any thoughts? (jan 2013)
by aademattia＠comcast.net 22 Apr '14

22 Apr '14

I use robodialer.net for our dialer traffic. They don't sell DID but good for dialer traffic. ----- Reply message ----- From: "naftula" <6279611(a)gmail.com> To: <freeswitch-users(a)lists.freeswitch.org> Subject: [Freeswitch-users] DID providers - any thoughts? (jan 2013) Date: Tue, Apr 22, 2014 1:40 PM Flowrout in my opinion is not an enterprise carrier-grade SIP trunk provider, they are a business consumer provider, the same is with vitality, they are a reseller of someone, I am looking for a real Enterprise provider, I have an option between 'Broadvox' and 'peerless network' does anyone have information on them? -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/DID-providers-any-thoughts-ja… Sent from the freeswitch-users mailing list archive at Nabble.com. _________________________________________________________________________ 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

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Re: [Freeswitch-users] DID providers - any thoughts? (jan 2013)
by Alex Kinch 22 Apr '14

22 Apr '14

We (Ziron) do SMS-enabled US DIDs, Canada coming soon... ping me off list ( alex.kinch(a)ziron.net) if you want more info or head over to ziron.net and signup for an account. ps: Cal - clearly not sending you enough brown envelopes to get a recommendation! :p On 22 April 2014 15:32, <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. transfer registration (Helena Garcia-Nieto) > 2. Invalid and unreliable DTMF (Manish Kumar) > 3. Re: DID providers - any thoughts? (jan 2013) > (Cal Leeming [Simplicity Media Ltd]) > 4. Re: SSL3_READ_BYTES:sslv3 alert handshake failure (Steven Ayre) > > > ---------- Forwarded message ---------- > From: "Helena Garcia-Nieto" <helena.gnieto(a)morodo.co.uk> > To: "'FreeSWITCH Users Help'" <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Tue, 22 Apr 2014 15:24:25 +0200 > Subject: [Freeswitch-users] transfer registration > > Hi all, > > > > I'm trying to transfer the registration to a gw. I want freeswitch to send > the registration message received from the external interface (phone via > internet) to the internal interface (to the internal gw, private interface) > Could someone point me in the direction of how to do it? I am reading sofia > module and some examples but I cannot find the right way. > > > > Thanks! > > > > Helena > > > ---------- Forwarded message ---------- > From: Manish Kumar <manish.kumar.fs(a)gmail.com> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Tue, 22 Apr 2014 19:20:16 +0530 > Subject: [Freeswitch-users] Invalid and unreliable DTMF > Hi, > > I am facing an issue where DTMF digits are wrongly captured or getting > entirely missed from being detected. > > Environment: > *Sangoma *A116 / A108 > *Freeswitch Version 1.2.14* > > *mod_freetdm* > perl ivr scripts using *perl ESL* > PRI lines from Indian operator. > > However the same script and same configuration works perfectly well on > Freeswitch *version 1.0.7*. Also none of the versions gave any issue when > i test by simply using mod_sofia and an extension on x-lite. > > I viewed the entire change logs from version 1.0.7 till 1.2.14, there are > no changes on DTMF since 1.0.7 > > Are there any possibilities that telco plays a role as well in such DTMF > issues.? > > > Thanks, > > > > > ---------- Forwarded message ---------- > From: "Cal Leeming [Simplicity Media Ltd]" < > cal.leeming(a)simplicitymedialtd.co.uk> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Tue, 22 Apr 2014 14:58:52 +0100 > Subject: Re: [Freeswitch-users] DID providers - any thoughts? (jan 2013) > > > > On Tue, Apr 22, 2014 at 2:22 AM, Brian West <brian(a)freeswitch.org> wrote: > >> Comments inline. >> >> On Apr 21, 2014, at 5:17 PM, Sean Devoy <sdevoy(a)bizfocused.com> wrote: >> >> > Hi Natula, >> > >> > I am ignoring who may or may not be supporters of FreeSwitch for this >> post. However, I do NOT think that should be ignored when making your >> selection and support for this product is very important in the long run. >> You can add that in to your decision after reviewing who meets your needs. >> > >> > I have been VERY pleased with VOIP service level and tech support from >> www.vitelity.com. There wholesales pricing is quite competitive >> although I have not shopped around in a year or so. Whole DIDs are >> $0.49/month, Caller ID service $1/month, E911 $1/month, typical WHOLESALE >> US calls prices 0.003 per minute (.3 cents). As full disclosure, I was NOT >> thrilled with their LNP department in the past, but here again have not >> ported much lately. I lease a server in their facility with FreeSwitch on >> it at an excellent price. The server lease applies to the wholesale >> monthly minimum. The server support is absolutely outstanding. >> >> No t.38, But they do have SMS that works with mod_dingaling (this was >> demoed at ClueCon last year) >> >> > >> > I have used www.Voipinnovations.com in the past. Their VOIP Tech >> support may be the best I ever used, especially LNPs. In many ways I wish >> I had stayed there, but I am happy enough with Vitelity to not expend the >> effort to switch back. The VOIP Service provided was also very good. >> >> Never heard of these guys. >> >> > >> > I have used DIDLOGIC, especially as a backup when calls fail to route. >> However, the sheer number login disconnects and reconnects in the log >> caused me to drop them. Others may have more recent experience. I have no >> HARSH things to say, just nagging little stuff I never worked out. >> >> Or these guys. >> > > Avoid. > > >> >> > >> > I still dabble with VOIP.MS. They have seemed GREAT every time I >> looked, except they did not offer E911 service. I understand they do now, >> but again I am settled in at Vitelity so I keep everything at one place. >> They should certainly be looked over. >> >> These guys seem ok too. >> >> > >> > Hope that helps. >> > Sean Devoy >> >> >> I use Flowroute, e911, CNAM Publish and dips for inbound, T.38 works >> really well. No SMS yet. *MO* WHERE ART THOU SMS!?!? >> > > +1 for flowroute, I've personally never used them but they come highly > rated. > > Clickatell for SMS. > > >> >> -- >> Brian West >> brian(a)freeswitch.org >> FreeSWITCH Solutions, LLC >> PO BOX 2531 >> Brookfield, WI 53008-2531 >> Twitter: @FreeSWITCH , @briankwest >> http://www.freeswitchbook.com >> http://www.freeswitchcookbook.com >> >> T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST >> iNUM: +883 5100 1420 9001 >> ISN: 410*543 >> Skype:briankwest >> PGP Key: http://www.bkw.org:81/key.txt (AB93356707C76CED) >> >> >> >> >> >> >> >> >> >> >> >> >> >> _________________________________________________________________________ >> 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: Steven Ayre <steveayre(a)gmail.com> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Tue, 22 Apr 2014 15:31:55 +0100 > Subject: Re: [Freeswitch-users] SSL3_READ_BYTES:sslv3 alert handshake > failure > Can't help you with what the issue would be (though I'd verify your > date/time settings are correct)... but I would update your OpenSSL version > since 1.0.1e is vulnerable to the heartbleed bug. > > > On 22 April 2014 13:46, Assaf Dahary <adahary(a)gmail.com> wrote: > >> Hi, >> >> >> >> I've successfully installed a FS server with TLS using PsitiveSSL and it >> is working great. >> >> >> >> Few days ago I've followed the same installation on another standalone >> machine with the same FS-1.2.22 and PsitivieSSL CA but this time I cannot >> connect over TLS. >> >> >> >> It seems that FS has no cipher to response with and it fails on >> negotiations. >> >> >> >> The PositiveSSL is OK because I verified it locally with "openssl >> s_client" and from the internet using browser/https. >> >> >> >> My ssl/ pem files are made with (like I did with the first server - OK): >> >> #cat mysite_com.crt myserver.key > agent.pem >> >> #cat PositiveSSLCA2.crt AddTrustExternalCARoot.crt > cafile.pem >> >> # chown freeswitch.freeswitch *.pem >> >> #chmod 640 *.pem >> >> >> >> When issuing "$ sslscan myfs.com:5091 | grep Accepted " >> >> I get no single cipher. I get long list of 'Rejected' ciphers. >> >> When I'm running the same command for my first server I get a list of supported ciphers - which is OK. >> >> >> >> When >> >> [root@www ~]# openssl s_client -connect myfs.com:5091 >> >> CONNECTED(00000003) >> >> depth=2 C = SE, O = AddTrust AB, OU = AddTrust External TTP Network, CN = AddTrust External CA Root >> >> verify return:1 >> >> depth=1 C = GB, ST = Greater Manchester, L = Salford, O = COMODO CA Limited, CN = PositiveSSL CA 2 >> >> verify return:1 >> >> depth=0 OU = Domain Control Validated, OU = PositiveSSL, CN = myfs.com >> >> verify return:1 >> >> 140160541112136:error:14094410:SSL routines:SSL3_READ_BYTES:sslv3 alert handshake failure:s3_pkt.c:1256:SSL alert number 40 >> >> 140160541112136:error:140790E5:SSL routines:SSL23_WRITE:ssl handshake failure:s23_lib.c:177: >> >> >> >> >> >> I've already re-installed FS with clean config files. >> >> >> >> Centos 6.x 64, OpenSSL 1.0.1e-fips 11 Feb 2013. >> >> >> >> I would appreciate any help/tip on this TLS fail issue. >> >> >> >> Regards >> >> >> >> Assaf >> >> >> >> >> _________________________________________________________________________ >> 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 > >

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SIP Client couldn't receive BYE
by Jimmy Chang 22 Apr '14

22 Apr '14

Hi, We have two different SIP Clients and they all use 3G to connect to FS. SIP_Client1 receives BYE without problem but SIP_Client2 couldn't. In the SIP_Client2 case, I don't understand why FS sended BYE to 100.77.163.183 rather than 42.71.215.54. Here are the snippet of FS log. http://pastebin.freeswitch.org/22430 Thanks. Jimmy --- 由於 avast! 防毒軟體已開啟防護，所以此封電子郵件將不受病毒及惡意軟體的侵害。 http://www.avast.com

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ClueCon Weekly News and Notes
by Ken Rice 22 Apr '14

22 Apr '14

Hey FreeSWITCHers, Its getting to be that time of the week again... When: Wed 1PM Eastern, 10AM Pacific, 1700 GMT Where: sip:888@conference.freeswitch.org +1 919 386 9900 http://wiki.freeswitch.org/wiki/Weekly_Conference_Call_Calling_Instructions for more info This week: Jeff AKA krzee on irc from Liberty Private Network will be talking about secure darknet voip service. Future Events: have any ideas on what you want to see? Email me! Do you have a contact with someone working on a cool project? Put them in touch with me! We would love to have them on! Also don¹t forget to get your ClueCon Registration in @ http://www.cluecon.com. Register by the end of the month with CLUECON2013 as the discount code and save $100! -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH

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Is there an (Overhead) "Page" Application/Function?
by Sean Devoy 22 Apr '14

22 Apr '14

Hi, If this exists and I have missed, I am sorry. However searching for "page" hits like every "Page n" in the wiki. I was demonstrating Freeswitch to a new customer who was rightfully impressed. As no good deed goes unpunished, they have a request. In this case "they" is the one who pays the bill! I was demonstrating "intercom" (aka auto-answer) and they were amazed. But, then HE said so is there a number I can dial so I can make announcements to EVERYBODY on their intercom? So I can brute force it in the dial plan, but before I jumped in, "isn't there an app for that?" The key features would be it sets auto-answer=true and dials everyone ELSEs extension, but not the one dialing. So, do I have to have an extension for this "feature" with nested <conditions> for caller id and bridge statements for each extension. This will of course have to be updated whenever an extension is added. That's not unreasonable for this customer as there are out of rooms to put people in. I was hoping to not re-invent the wheel. Also, if someone had a cute script to share that would be awesome. Maybe something that identified all currently registered users (in this REALM) and built the bridge statement dynamically, excluding the caller??? I would love to start learning more of this type of scripting function. Thanks in advance, Sean

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python ESL package for debian
by Anish Patel 22 Apr '14

22 Apr '14

Hello All, I was wondering if there is a python ESL package for debian or if its in pypi somewhere. I've dug around a bit and can't seem to locate it, and am currently just building it from the sources. Thanks in advance, Anish

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Gateway configuration
by Helena Garcia-Nieto 22 Apr '14

22 Apr '14

Hi All! I am starting with freeswitch. I decided to test basic stuff before configuring any further. I am defining a simple gateway (see config file) and using it to transfer any call starting with 00. The config I've seen and the emails I've checked say the correct config should be: <extension name="mycall_extensions"> <condition field="destination_number" expression="^00(\d{3,20})$"> <action application="log" data="INFO Inside MyCall extensions --$1_"/> <action application="bridge" data="sofia/external/mycall/$1"/> </condition> </extension> But if I use it like this I got the error: 2014-04-15 14:49:44.289151 [INFO] mod_dialplan_xml.c:558 Processing HelenaGN <1001>->00aaaaaa in context public 2014-04-15 14:49:44.289151 [INFO] mod_dptools.c:1595 Inside MO-Call extensions --aaaaaa_ 2014-04-15 14:49:44.289151 [WARNING] mod_sofia.c:5082 Cannot locate registered user mycall/aaaaaaaa@external 2014-04-15 14:49:44.289151 [NOTICE] mod_sofia.c:5339 Close Channel N/A [CS_NEW] 2014-04-15 14:49:44.289151 [NOTICE] switch_ivr_originate.c:2670 Cannot create outgoing channel of type [sofia] cause: [USER_NOT_REGISTERED] 2014-04-15 14:49:44.289151 [INFO] mod_dptools.c:3201 Originate Failed. Cause: USER_NOT_REGISTERED 2014-04-15 14:49:44.289151 [NOTICE] switch_channel.c:4646 Hangup sofia/external/1001@xx.xx.xx.xx:5080 [CS_EXECUTE] [USER_NOT_REGISTERED] But. Why is it looking for the aaaa number on the external instead of the mycall? If I configure it with <action application="bridge" data="sofia/external/mycall/$1(a)mydomain.realm"/> It finds its way to the gateway. I don't really understand the point of defining the whole gw if you need to specify the realm here again anyway. Could someone point me in the direction of why the basic config did not work? Thanks Helena clude> <gateway name="mycall"> <param name="realm" value="mydomain.realm "/> <param name="username" value="notimportant"/> <param name="password" value="notimportant"/> <param name="register" value="false"/> </gateway> </include>

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Docs call tomorrow April 22nd at 2PM Central in 888
by Brian West 22 Apr '14

22 Apr '14

FreeSWITCHers, Docs meet up tomorrow, 2PM Central. We’ll touch base and go over our progress and verify our approach is correct. Anyone wishing to help with the Confluence migration please feel free to call in tomorrow. I’ll probably be in 888 sooner and probably most of tomorrow if anyone else has any questions, or other issues they wish to discuss. See you there… -- Brian West brian(a)freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org:81/key.txt (AB93356707C76CED)

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