- FreeSWITCH-users - FreeSWITCH Mailing Lists

Issues with Linksys ATA and DTMF
by obbyone 19 Jan '12

19 Jan '12

Hi, On my freeswitch server, I have created a number 9910 that has to recieve messages in DTMF. As I call that number with a softphone (xlite) on a PC, everything works, but as I try to call and send DTMF from an analog phone plugged into an ATA linksys spa 3000, freeswitch answers, but no DTMF is recieved. The linksys is configured to send DTMF in inband mode. Do you have any idea how I can handle this issue ? Thanks, Tony -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Issues-with-Linksys-ATA-and-D… Sent from the freeswitch-users mailing list archive at Nabble.com.

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Freeswitch & double NAT Configuration (Was: USER_NOT_REGISTERED with external profile)
by Alexis de BRUYN [Mailinglists] 19 Jan '12

19 Jan '12

Hi Everybody, I am trying to get a conversation between two natted clients and a natted freeswitch server. I am playing with a fresh install, the external profile and default directory (1000 & 1001 users). I can see that my clients are registered on the server logs (on the 5080 port). All others needed ports are opened too. I have followed the instructions from the wiki to get a double nat configuration, but my client phone are not ringing (and I cannot hear anything). All is working fine with defaults in a non nat configuration (with internal profile). Is there anything special to set up in the directory / user configuration ? Can anyone give me some hints with his natted configuration ? I am stuck. Thanks for your help. Regards, -------- Original Message -------- Subject: [Freeswitch-users] USER_NOT_REGISTERED with external profile Date: Mon, 16 Jan 2012 23:37:20 +0100 From: Alexis de BRUYN [Mailinglists] <alexis.mailinglist(a)de-bruyn.fr> Reply-To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> To: freeswitch-users(a)lists.freeswitch.org Hi Everybody, I am trying to use FreeSwitch in a (double) NAT Configuration from a fresh snapshot install (Debian Squeeze, server outside from my client LAN) with default directory for users. When I call 1001 from 1000 (or 1000 from 1001) , I cannot hear the callee, the phone doesn't ring and automatically hangup, I see in the console : 2012-01-16 22:59:37.691663 [ERR] switch_ivr_originate.c:2447 Cannot create outgoing channel of type [error] cause: [USER_NOT_REGISTERED] 2012-01-16 22:59:37.691663 [ERR] switch_ivr_originate.c:2447 Cannot create outgoing channel of type [user] cause: [USER_NOT_REGISTERED] However, 1000 and 1001 are registered (from the same LAN) : sofia status profile external ================================================================================================= Name external Domain Name N/A Auto-NAT false DBName sofia_reg_external Pres Hosts Dialplan XML Context public Challenge Realm auto_to RTP-IP 192.168.1.6 SIP-IP 192.168.1.6 URL sip:mod_sofia@192.168.1.6:5080 BIND-URL sip:mod_sofia@192.168.1.6:5080 HOLD-MUSIC local_stream://moh OUTBOUND-PROXY N/A CODECS IN G7221@32000h,G7221@16000h,G722,PCMU,PCMA,GSM CODECS OUT PCMU,PCMA,GSM TEL-EVENT 101 DTMF-MODE rfc2833 CNG 13 SESSION-TO 0 MAX-DIALOG 0 NOMEDIA false LATE-NEG false PROXY-MEDIA false AGGRESSIVENAT false STUN-ENABLED true STUN-AUTO-DISABLE false CALLS-IN 3 FAILED-CALLS-IN 0 CALLS-OUT 0 FAILED-CALLS-OUT 0 Registrations: ================================================================================================= Call-ID: aU.1ZlEZkW7nlHpwp9ig1lt7A3weNnwm User: 1001(a)X.Y.Z.T Contact: "Freeswitch" <sip:1001@A.B.C.D:50193> Agent: Bria iOS 2.0.0 Status: Registered(UDP)(unknown) EXP(2012-01-16 23:16:00) EXPSECS(934) Host: phone IP: A.B.C.D Port: 50193 Auth-User: 1001 Auth-Realm: X.Y.Z.T MWI-Account: 1001(a)X.Y.Z.T Call-ID: 3c27b7424837-bsfi877ujb2n User: 1000(a)X.Y.Z.T Contact: "freeswitch" <sip:1000@192.168.0.69:2048;line=skgk4hug> Agent: snom300/8.4.32 Status: Registered(UDP)(unknown) EXP(2012-01-17 00:01:24) EXPSECS(3658) Host: phone IP: A.B.C.D Port: 62061 Auth-User: 1000 Auth-Realm: X.Y.Z.T MWI-Account: 1000(a)X.Y.Z.T Total items returned: 2 ================================================================================================= All necessary ports are opened/forwarded on the server. I See on the 1000 configuration that this is the local ip address which is set as contact. Is there any other setups to do in the directory ? Or other parameters in the external profile ? Thanks for your help ! Regards, -- Alexis de BRUYN Mail : alexis.mailinglist(a)de-bruyn.fr _________________________________________________________________________ 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 -- -- Alexis de BRUYN Mail : alexis.mailinglist(a)de-bruyn.fr

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feature list
by bob smith 19 Jan '12

19 Jan '12

Hi everyone. Does anyone have a feature list for freeswitch? I have not been able to find a good one anywhere ! Pls let me know ! B

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Unable to export a variable to b leg (callcenter)
by Yungwei Chen 19 Jan '12

19 Jan '12

Hi, With the extension below, I am able to export caller_id_name to b leg without problems. However, exporting another variable doesn't seem to work because I don't see it in the corresponding CDRs (cdr.variables). It looks like only known variables are exportable. Is there way to export a custom variable in this case? Thanks. <extension name="test"> <condition field="destination_number" expression="^test1$" break="on-true"> <action application="export" data="sid=${sid}" />  <action application="export" data="caller_id_name=${origination_caller_id_name}" />  <action application="answer" data=""/> <action application="callcenter" data="support@default"/> </condition> </extension>

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choppy sound on voice recording and conference
by Anestis Mavro 18 Jan '12

18 Jan '12

Hello, I have a problem with one of my sip trunks. The provider uses g729 with a ptime of 40ms and all incoming calls that are being recorded or go to the conference have a big delay and choppy sound, like a mismatch of ptime. I have already tried to put rtp-autofix-timing=false into the profiles and even changed the inbound-codec.. to g729@40i, but nothing helps. We have done a test with the provider switching for a test call to 20ms and the sound was perfect. The issue now is that the provider can't keep this setting only for us, he wants to send with 40ms. Is there a way to configure it on Freeswitch? Thank you Anestis

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mod_sofia displaying on Avaya 9630 as caller id contact
by Bernard Murphy 18 Jan '12

18 Jan '12

Hi, I have a soft phone calling an Avaya 9630 both registered on the same freeswitch and have a problem with the Avaya displaying the inbound contact as 'mod_sofia' instead of the originating extension number. The invite from the originating extension and the invite to the Avaya are both pasted below and you will see theat freeswitch includes mod_sofia as the contact uri. Can this be corrected in anyway so that the Avaya phone displays the incoming extension number ? thanks murph* ORIGINAL INVITE* INVITE sip:1000@192.168.2.42 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.31;rport;branch=z9hG4bKc0a8021f000000234f166fd50000386400000044 Content-Length: 337 Contact: <sip:1001@192.168.2.31:5060> Call-ID: BC96029E-D214-4CBF-BCF5-BD0A260757C1(a)192.168.2.31 Content-Type: application/sdp CSeq: 2 INVITE From: "unknown"<sip:1001@192.168.2.42>;tag=1383828710312 Max-Forwards: 70 To: <sip:1000@192.168.2.42> User-Agent: SJphone/1.60.289a (SJ Labs) Proxy-Authorization: Digest username="1001",realm="192.168.2.42",nonce="6ff3bebc-b5c5-446a-8625-e89dcd2dd861",uri=" sip:1000@192.168.2.42 ",response="1754cba9b8bc72cfac8c69ff0bee8a21",algorithm="MD5",cnonce="1383887920463",qop="auth",nc="00000001" *INVITE FROM SOFIA* INVITE sip:1000@192.168.2.30;avaya-sc-enabled;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.2.42;rport;branch=z9hG4bKjtc19v5754K6c Max-Forwards: 69 From: "Extension 1001" <sip:1001@192.168.2.42>;tag=QvNQrZya6vHjS To: <sip:1000@192.168.2.30;avaya-sc-enabled;transport=udp> Call-ID: f73b031a-bc45-122f-388f-000c291b7d13 CSeq: 23099169 INVITE Contact: <sip:*mod_sofia*@192.168.2.42:5060> User-Agent: FreeSWITCH-mod_sofia/1.0.6-hacked-20120115T023659Z Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 427 X-FS-Support: update_display Remote-Party-ID: "Extension 1001" <sip:1001@192.168.2.42 >;party=calling;screen=yes;privacy=off

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Re: [Freeswitch-users] record_fsv buffers - Any way to flush incoming video?
by Greg Millam 18 Jan '12

18 Jan '12

> commit c358f67fe4348b8b5209328660aef02d8ffaf15f > Author: Anthony Minessale <anthm(a)freeswitch.org> > Date: Tue Jan 17 12:19:23 2012 -0600 > > eat inbound vid while playing fsv files > > > I didn't have a way to test it setup so i'll need feedback. It works, but now since it's missing the first frame, the resulting video is only partial, showing new bits. Is there a way to send a message to the client to "clear and send a full frame of video" ? > On Mon, Jan 16, 2012 at 11:38 PM, Greg Millam <greg(a)brilliantecho.com> wrote: >> >> Hi folks - >> >> I have a freeswitch dialplan + script that first calls play_fsv to >> play a greeting, then record_fsv to record incoming video to fsv. >> >> It works fine, but there's one issue: Apparently, freeswitch is >> buffering incoming video during play_fsv. When record_fsv is called, >> that buffer is dumped into the .fsv file, resulting in several seconds >> of unneeded video. >> >> Is there a way to empty that incoming video buffer before record_fsv >> begins recording? >> >> Thank you! >> >> - Greg Millam

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GSMopen and Embedded Combined Devices
by Komar, Jason 18 Jan '12

18 Jan '12

Hi, I'm thinking about giving GSMopen a try and am wondering if anyone has a good source and recommendation for an embedded combined device for use with it? I have done some web searching, but must not be using the best search terms. Thanks, Jason Komar

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Variable in dialplan
by Miha Zoubek 18 Jan '12

18 Jan '12

Hi, I need a little help about passing variable between dialplans. I have created one default diaplan. In this dialplan a have set: <action application="execute_extension" data="check special numbers XML default"/> In check special numbers XML there is a condition (<condition field="destination_number" expression="^check special numbers$" >) and than <action application="execute_extension" data="special numbers XML default"/>. In special numbers XML I have a condition and if this condition is right the variable $test is set to 112. SO in default dialplan after all this is set must default dialplan execute this <action application="bridge" data="sofia/external/386${test}(a)xx.xxx.xxx.xxx"/> where ${test} should be variable from special numbers XML. Why the value of variable is not passed? Thanks! -- Best regards / Lep Pozdrav Miha Zoubek Softnet d.o.o.

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Question about the future of mod_nibblebill
by Carlo Dimaggio 18 Jan '12

18 Jan '12

Hi all, I'm thinking about a development of a custom prepaid application based on mod_nibblebill (that is very interesting). I'm wondering about the status and the future of this module in order to understand if this is the better way to achieve the aim of the project. I have seen in the mod_nibblebill source code this notes: * TODO: Fix what happens when the DB is not available * TODO: Fix what happens when the DB queries fail (right now, all are acting like success) * TODO: Add buffering abilities * TODO: Make error handling for database, such that when the database is down (or not installed) we just log to a text file * FUTURE: Possibly make the hooks not tied per-channel, and instead just do this as a supervision style application with one thread that watches all calls Does the developers think to continue supporting the module (not only bugfix but new features)? I'm sorry for the particular question but I wouldn't do errors at this stage of the project. Best Regards,

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