January 2015 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Re: [Freeswitch-users] Cisco 7942 with freeswitch - Multitenant Environment
by Luis Daniel Lucio Quiroz 03 Jan '15

03 Jan '15

Are you using the fqdn instead the ip?? On Jan 3, 2015 11:00 AM, "Akhil Chandra" <achandra(a)telrouter.com> wrote: > Hi Users, > > Trying to get Cisco 7942 working with freeswitch multitenant environment. > > The issue I am having is that Cisco 7942 tries to register with > username@<IP addresss of freeswitch> rather than username@<domain name > of freeswitch> .. > > I am using Cisco 7942 latest version ( 9.4) .. The config files have all > domain names in them .. > > > Is there a setting in the config file that I am missing for the Cisco to > send the domain name instead of the IP address. > > Any insights appreciated. > > Regards, > Akhil > > > Sent from Windows Mail > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting(a)freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.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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Cisco 7942 with freeswitch - Multitenant Environment
by Akhil Chandra 03 Jan '15

03 Jan '15

Hi Users, Trying to get Cisco 7942 working with freeswitch multitenant environment. The issue I am having is that Cisco 7942 tries to register with username@<IP addresss of freeswitch> rather than username@<domain name of freeswitch> .. I am using Cisco 7942 latest version ( 9.4) .. The config files have all domain names in them .. Is there a setting in the config file that I am missing for the Cisco to send the domain name instead of the IP address. Any insights appreciated. Regards, Akhil Sent from Windows Mail

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problem with sequential calling & mod lua session originate
by spros p 03 Jan '15

03 Jan '15

Hello, I am trying to originate a series of calls in sequential order. I am using the pipe notation, for the dial string, however the call never reaches the second destination. When trying a comma or when using the enterprise originate notation (:_:), calls are indeed simultaneously placed. The session is being called as follows in LUA: origination_string = "{ignore_early_media=true,originate_timeout=30,continue_on_fail=true}[origination_caller_id_number=XXXX]sofia/gateway/provider/AAAAAA|[origination_caller_id_number=XXXX]sofia/gateway/provider/BBBBBB" new_session = freeswitch.Session(origination_string, session); Any help would be greatly appreciated. Many Thanks, S

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Re: [Freeswitch-users] Canceling sched_hangup and UUID lookup
by DanB 03 Jan '15

03 Jan '15

Hi Kurtis, Interesting approach you are trying to take. I see however some scalability issues. Unless you are planning to build an office PBX (for which mod_nibblebill should do just fine) and want to go for a more scalable solution, you will be forced to send calls belonging to the same account always to the same node in order to keep the billing for them. Regarding your float64 comments, you are right about trying to avoid them but there it depends on the operations you are doing on the numbers (eg: sum/multiply vs divide). Here is how we solved in CGRateS the issues you are facing: * Each call started will have the global session timeout set on call setup stage (same as you planned it). This is more like security for your call timeout. * We keep an external component responsible for call management and timeout corrections. The timeouts will be recalculated inside external component and if it will be necessary, the sessions will be killed via ESL. This solves another issue you did not cover yet: if you are billing units in advance, you can also refund non-used units at the call end. * On the float issue, we are passing all the calculations through a special object which does the rounding where needed, eliminating the float issues with calculations. So far we could not find a case failing, although we considered initially converting currencies internally to huge-INTs and perform all the operations there (this would force us using always same unit of currency - which we were trying to avoid). Hope it helps! DanB

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SSD Drives vs 15k drives
by Andrew 02 Jan '15

02 Jan '15

Hi, We are going to be buying new servers to run FS as a sip server. I know some people say use SSD drives but I wanted to ask. Performance is a big deal so I like to do anything I can to even get one more CPS. We will be running the FS with no SQL and no RTP(media bypass) and in a ram drive. Will the SSD Drives even be hit in this case? Andrew

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Designing a schema for CDRs
by Tayyab Atiq 02 Jan '15

02 Jan '15

Hi. I have recently been trying to process CDR from mod_xml_cdr and store it in a db I first parsed and wrote all the information in a file, then designed a database based on the fields that appeared in the file. But it seems that FS adds/removes fields sometimes. This causes my query to fail as my program tries to insert columns that do not exist. Is there a way to figure out all the possible fields that might appear in the xml sent by mod_xml_cdr?

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Re: [Freeswitch-users] Designing a schema for CDRs
by Dan-Cristian Bogos 02 Jan '15

02 Jan '15

Hi Tayyab Another option you can have a look at is the CDR server integrated within CGRateS. You are not forced to any business logic (once stored the CDRs will be queriable/exportable over JSON APIs). You can use or not the rating which comes as bonus in this case and fields which go in are selectable in config templates. DanB. Am 02.01.2015 19:44 schrieb <freeswitch-users-request(a)lists.freeswitch.org>: > 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: Control of IPv6 vs. IPv4 (Brian West) > 2. Re: how can i stop freeswitch external profile listening on > tcp port (Aqs Younas) > 3. Re: Designing a schema for CDRs (tayyabatiq(a)gmail.com) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Fri, 2 Jan 2015 11:13:01 -0600 > From: Brian West <brian(a)freeswitch.org> > Subject: Re: [Freeswitch-users] Control of IPv6 vs. IPv4 > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Message-ID: > <CAEJMVkAvvyFtM7G0j7F-Z9= > RxtBRt4RfR50JAkN9D85+8L-Jyw(a)mail.gmail.com> > Content-Type: text/plain; charset="utf-8" > > FreeSWITCH does one or the other, you should pick where you want your > gateway to register at, and only allow it there and there only. > > On Fri, Jan 2, 2015 at 10:02 AM, George F. Phelps < > GeorgePhelps(a)gfphelps.com > > wrote: > > > My VoIP provider supports both IPv4 and IPV6 registrations. I can only > > have one registration active at a time. Freeswitch is attempting both > IPv4 > > and IPv6 connections. Randomly, sometimes a IPv4 connection is the first > > (only) registration established; other times it is the IPv6 connection. > > > > > > > > How to I configure Freeswitch to deterministically only attempt one type > > (my choice of either IPv4 or IPv6) of connection? > > > > > > > > freeswitch(a)ip-172-31-33-109.ec2.internal> sofia status > > > > > > > > Name > > Type Data State > > > > > > > ================================================================================================= > > > > external-ipv6 profile sip:mod_sofia@ > [::1]:5080 > > RUNNING (0) > > > > external-ipv6::switch2voip.us gateway > > sip:XXXXXXXXXX@66.33.147.150 REGED > > > > external profile > > sip:mod_sofia@54.174.255.168:5080 RUNNING (0) > > > > external::switch2voip.us gateway > > sip:XXXXXXXXXX@66.33.147.150 TRYING (retry: 20s) > > > > 172.31.33.109 alias > > internal ALIASED > > > > internal-ipv6 profile sip:mod_sofia@ > [::1]:5060 > > RUNNING (0) > > > > internal profile > > sip:mod_sofia@54.174.255.168:5060 RUNNING (0) > > > > > > > ================================================================================================= > > > > 4 profiles 1 alias > > > > > > > > freeswitch(a)ip-172-31-33-109.ec2.internal> > > > > > > > > Thanks, > > > > > > > > George > > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting(a)freeswitch.org > > http://www.freeswitchsolutions.com > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://confluence.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 > > > > > > -- > > *Brian West* > brian(a)freeswitch.org > > > *Twitter: @FreeSWITCH , @briankwest* > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) > *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest >

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How to transfer call from one context to another after a certain time
by Aqs Younas 02 Jan '15

02 Jan '15

Hi, Users. I want to transfer my call from one context public(which is playing a never ending stream with mod_shout) to another context Second (which is also playing a never ending stream) after a certain time like, after 30 mints, without making user press any dtmf. <include> <context name="public"> <extension name="First_Stream"> <condition field="destination_number" expression="^750$"> <action application="answer"/> <action application="playback" data="shout:// scfire-dll-aa02.stream.aol.com:80/stream/1074"/> <action application="transfer" data="${destination_number} XML Second"/> </condition> </extension> </context> <context name="Second"> <extension name="Second_Stream"> <condition field="destination_number"> <action application="playback" data="shout:// scfirl-bb1.stream.aol.com:80/stream/1870"/> </condition> </extension> </context> </include> How can i do so? I can write above dial plan with lua/python but don't know how to transfer call without making user press any key. Since "play_and_get_digits" and "read" command also need some dtmf for next line execution. Many Thanks

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how can i stop freeswitch external profile listening on tcp port
by Aqs Younas 02 Jan '15

02 Jan '15

Hello, I am newbie to freeswitch. When i run root@66-226-74-508:~# netstat -nlp | grep freeswitch tcp 0 0 206.225.85.25:6020 0.0.0.0:* LISTEN 4556/freeswitch udp 0 0 206.225.85.25:6020 0.0.0.0:* 4556/freeswitch i see freeswitch external profile listening over tcp and udp port. But i only want external profile to listen over udp port. How can i do so? Thanks

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zrtp sas1
by Henk Dick 02 Jan '15

02 Jan '15

Who has a suggestion? I have the following setup: jitsi --- freeswitch --- ata --- dect I have the jitsi client working correctly running with zrtp. Meaning that I will get sas1 shown at softhone the first time I make a call or when I have removed the cache file db/zrtp.dat De second call will not show the sas1. In the lua script do I want to use say to announce the sas1 I get from freeswitch. I am getting in the script the following vars. ZRTP User first time: 2015-01-02 14:24:23.938928 [INFO] switch_cpp.cpp:1288 ZRTP Session Settings: 2015-01-02 14:24:23.938928 [INFO] switch_cpp.cpp:1288 zrtp_secure_media_confirmed:true 2015-01-02 14:24:23.938928 [INFO] switch_cpp.cpp:1288 zrtp_new_user_enrolled:true 2015-01-02 14:24:23.938928 [INFO] switch_cpp.cpp:1288 zrtp_already_enrolled:not_defined 2015-01-02 14:24:23.938928 [INFO] switch_cpp.cpp:1288 zrtp_sas1_string:w5bd 2015-01-02 14:24:23.938928 [INFO] switch_cpp.cpp:1288 zrtp_sas2_string:not_defined 2015-01-02 14:24:23.938928 [INFO] switch_cpp.cpp:1288 zrtp_enroll_complete:not_defined 2015-01-02 14:24:23.938928 [INFO] switch_cpp.cpp:1288 zrtp_user_unenrolled:true I expected that I could use them to determine when I have to announce the sas1 but unfortunately have the vars the same values with the second and the first call. Does anyone have a suggestion how to determine that the sas1 should be announced? Best regards Henk - osocoms

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