April 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

ClueCon 2012 Preview: Vestec Automatic Speech Recognition
by Michael Collins 30 Apr '12

30 Apr '12

Greetings! This week we have a small taste of ClueCon 2012: The Vestec <http://www.vestec.com?cc120430>team will be joining us on the FreeSWITCH community conference<http://wiki.freeswitch.org/wiki/Weekly_Conference_Call?cc12-0430>call this Wednesday, May 2. Vestec produces high-quality automatic speech recognition (ASR) software for open source telephony projects like Asterisk and FreeSWITCH. We look forward to having a Vestec engineer on the call with us this week to give us a technical look at their product and to answer community questions. Vestec will also be at ClueCon 2012 as a Silver Sponsor and have committed to being available for one-on-one consulting for those wanting to discuss their specific projects. Yet another reason to join us at ClueCon this August! The ClueCon 2012 speakers list is starting to take shape. We have a number of "fan-favorites" scheduled as well as a few new surprises. We are pleased to have returning for a third consecutive year Mr. Philip Zimmermann, inventor of PGP <http://en.wikipedia.org/wiki/PGP> and designer of the the ZRTP <http://zfone.com?cc12-0430> secure VoIP protocol. As we've previously announced <http://freeswitch.org/node/388?cc12-0430>, ZRTP is now included in the FreeSWITCH distribution. Philip will be discussing VoIP security in general, as well developments with the ZRTP project. Register <http://www.cluecon.com/register?cc120430> for ClueCon 2012 today and take advantage of ClueCon Power Play, which gives you extra entries in the world famous giveaways we have each year. Feel free to contact us if you have any questions about attending ClueCon, sponsoring, or speaking at the event. We look forward to seeing you there - in less than 100 days! -- Michael S Collins ClueCon Team http://www.cluecon.com 877-7-4ACLUE cc12-0430

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ClueCon 2012 Preview: Vestec Automatic Speech Recognition
by Michael Collins 30 Apr '12

30 Apr '12

Greetings! This week we have a small taste of ClueCon 2012: The Vestec <http://www.vestec.com?cc120430>team will be joining us on the FreeSWITCH community conference<http://wiki.freeswitch.org/wiki/Weekly_Conference_Call?cc12-0430>call this Wednesday, May 2. Vestec produces high-quality automatic speech recognition (ASR) software for open source telephony projects like Asterisk and FreeSWITCH. We look forward to having a Vestec engineer on the call with us this week to give us a technical look at their product and to answer community questions. Vestec will also be at ClueCon 2012 as a Silver Sponsor and have committed to being available for one-on-one consulting for those wanting to discuss their specific projects. Yet another reason to join us at ClueCon this August! The ClueCon 2012 speakers list is starting to take shape. We have a number of "fan-favorites" schedule as well as a few new surprises. We are pleased to have returning for a third consecutive year Mr. Philip Zimmermann, inventor of PGP <http://en.wikipedia.org/wiki/PGP> and designer of the the ZRTP <http://zfone.com?cc12-0430> secure VoIP protocol. As we've previously announced, ZRTP is now included in the FreeSWITCH distribution. Philip will be discussing VoIP security in general, as well developments with the ZRTP project. Register <http://www.cluecon.com/register?cc120430> for ClueCon 2012 today and take advantage of ClueCon Power Play, which gives you extra entries in the world famous giveaways we have each year. Feel free to contact us if you have any questions about attending ClueCon, sponsoring, or speaking at the event. We look forward to seeing you there - in less than 100 days! -- Michael S Collins ClueCon Team http://www.cluecon.com 877-7-4ACLUE cc12-0430

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Forward incoming call to cell phone if user not reachable/registered
by Travis Langhals 30 Apr '12

30 Apr '12

Could someone please help with an example dialplan for call failover to an external DID (i.e. cell phone or pots line) in case the the user has a network failure can cannot be reached?

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Freeswitch speaks italian
by Chris B. Ware 30 Apr '12

30 Apr '12

Hi, some time ago, Mr. Marco Menardi and his wife Paola (the voice) made a set of audio files for Asterisk in italian, under creative commons license. Using Audacity I rearranged them to be used with FS. Translated folders are: ascii currency digits phonetic-ascii time voicemail Only 8Khz version, sorry. Link to my blog, where you can download files: http://chrisbware.wordpress.com/2012/04/12/freeswitch-parla-italiano/

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unable to load module. mod_gsmopen.so ( Invalid ELF header )
by awais nazir 30 Apr '12

30 Apr '12

hi, i am unable to load module, when i try to load modules (mod_gsmopen) in freeswitch CLI. seem error like .... freeswitc->freeswitc->freeswitc-> load mod_gsmopen+OK Reloading XML -ERR [module load file routine returned an error]2012-04-30 15:34:14.843048 [CRIT] switch_loadable_module.c:1300 Error Loading module /usr/local/freeswitch/mod/mod_gsmopen.so **/usr/local/freeswitch/mod/mod_gsmopen.so: invalid ELF header** 2012-04-30 15:34:14.843048 [INFO] mod_enum.c:812 ENUM Reloaded freeswitc-> freeswitc-> !Please help

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Re: [Freeswitch-users] Time routing issue
by Chris B. Ware 30 Apr '12

30 Apr '12

Hi, thanks Jeff and Michael for your help. Of course I set my variable with inline="true". To explain better, I have a catch-all extensions that call a lua script to initialize a set of chan variables according to the called number (destination_number). The script is executed inline. After that I put a lot of application=log on my dialplan and I can see ${rmon} correctly set. One of my log apps is the anti-action on the time routing condition, so to check ${rmon} when it fails. And has the right value! I'm wondering if mon,wday,time-of-day can be match against a variable. I think I have to use reg-exp or lua to do it :(

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wrong detection dtmf rfc2833
by Lubimov Alexey 30 Apr '12

30 Apr '12

Good day! I have problem with dtmf on incoming calls. Internal dtmf work normal, but on incoming call i have many mistakes (111 or 121 instead 110, for example). siptrace wrong detection attached, Abonent send 110, but freeswitch detect 111. in profile's config i set these options. (Im trying many variants, but no find solution) <param name="dtmf-duration" value="2000"/> <param name="dtmf-type" value="rfc2833"/> <param name="rfc2833-pt" value="0"/> <param name="liberal-dtmf" value="true"/> Have you any ideas?

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Time routing issue
by Chris B. Ware 30 Apr '12

30 Apr '12

Hi, I'd like to set a channel variable and use it in a time condition like this: <condition mon="${rmon}" > with rmon=1-12 (just to be sure condition is verified). I can't find out why it doesn't work. - Using <action application="info"/> , I can see variable is set [1-12] - I've tried also <condition mon="${rmon}" inline="true"> - Using directly <condition mon="1-12" > it works What's wrong? Help, please

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rfc2833 first Marker Bit not set to True on bleg
by Bernard Murphy 29 Apr '12

29 Apr '12

On the very latest GIT and previous , Freeswitch doesn't seem to set the first Marker Bit to true on an rfc2833 event on the bleg of a call out to a gateway on the A-leg the first event from my UA to Freeswitch is correct with duration as 0 and Marker bit set to True. Then the second event increments the duration and Market bit is set to false. but on the B-leg from freeswitch to the cisco gateway the first event has duration 80 (may or may not be a problem) but the Marker bit is set to False which must be wrong ? Duration then increments and Marker bit remains false. I also tried <param name="manual-rtp-bugs" value="START_SEQ_AT_ZERO"/> to make the duration start at zero but it seemed to have no effect. *ALEG from SIP UA to FREESWITCH* No. Time Source Destination Protocol Length Info 5254 23 10.0.47.110 10.50.47.30 RTP EVENT 60 Payload type=RTP Event, DTMF One 1 Frame 5254: 60 bytes on wire (480 bits), 60 bytes captured (480 bits) Ethernet II, Src: Sonicwal_3d:2b:1c (00:06:b1:3d:2b:1c), Dst: Vmware_ab:60:ec (00:0c:29:ab:60:ec) Internet Protocol Version 4, Src: 10.0.47.110 (10.0.47.110), Dst: 10.50.47.30 (10.50.47.30) User Datagram Protocol, Src Port: 49194 (49194), Dst Port: 21374 (21374) Real-Time Transport Protocol [Stream setup by SDP (frame 1122)] [Setup frame: 1122] [Setup Method: SDP] 10.. .... = Version: RFC 1889 Version (2) ..0. .... = Padding: False ...0 .... = Extension: False .... 0000 = Contributing source identifiers count: 0 1... .... = Marker: True Payload type: telephone-event (101) Sequence number: 26364 [Extended sequence number: 91900] Timestamp: 133920 Synchronization Source identifier: 0x03bf0676 (62850678) RFC 2833 RTP Event Event ID: DTMF One 1 (1) 0... .... = End of Event: False .0.. .... = Reserved: False ..00 1010 = Volume: 10 Event Duration: 0 *ALEG SECOND EVENT FROM SIP UA TO FREESWITCH* No. Time Source Destination Protocol Length Info 5255 23 10.0.47.110 10.50.47.30 RTP EVENT 60 Payload type=RTP Event, DTMF One 1 Frame 5255: 60 bytes on wire (480 bits), 60 bytes captured (480 bits) Ethernet II, Src: Sonicwal_3d:2b:1c (00:06:b1:3d:2b:1c), Dst: Vmware_ab:60:ec (00:0c:29:ab:60:ec) Internet Protocol Version 4, Src: 10.0.47.110 (10.0.47.110), Dst: 10.50.47.30 (10.50.47.30) User Datagram Protocol, Src Port: 49194 (49194), Dst Port: 21374 (21374) Real-Time Transport Protocol [Stream setup by SDP (frame 1122)] [Setup frame: 1122] [Setup Method: SDP] 10.. .... = Version: RFC 1889 Version (2) ..0. .... = Padding: False ...0 .... = Extension: False .... 0000 = Contributing source identifiers count: 0 0... .... = Marker: False Payload type: telephone-event (101) Sequence number: 26365 [Extended sequence number: 91901] Timestamp: 133920 Synchronization Source identifier: 0x03bf0676 (62850678) RFC 2833 RTP Event Event ID: DTMF One 1 (1) 0... .... = End of Event: False .0.. .... = Reserved: False ..00 1010 = Volume: 10 Event Duration: 160 *BLEG FIRST EVENT FROM FREESWITCH* *TO CISCO* No. Time Source Destination Protocol Length Info 5329 23 10.50.47.30 10.50.47.19 RTP EVENT 58 Payload type=RTP Event, DTMF One 1 Frame 5329: 58 bytes on wire (464 bits), 58 bytes captured (464 bits) Ethernet II, Src: Vmware_ab:60:ec (00:0c:29:ab:60:ec), Dst: 64:9e:f3:11:ff:60 (64:9e:f3:11:ff:60) Internet Protocol Version 4, Src: 10.50.47.30 (10.50.47.30), Dst: 10.50.47.19 (10.50.47.19) User Datagram Protocol, Src Port: 24034 (24034), Dst Port: 18768 (18768) Real-Time Transport Protocol [Stream setup by SDP (frame 1113)] [Setup frame: 1113] [Setup Method: SDP] 10.. .... = Version: RFC 1889 Version (2) ..0. .... = Padding: False ...0 .... = Extension: False .... 0000 = Contributing source identifiers count: 0 0... .... = Marker: False Payload type: telephone-event (101) Sequence number: 24716 [Extended sequence number: 90252] Timestamp: 158240 Synchronization Source identifier: 0x57e5d7a1 (1474680737) RFC 2833 RTP Event Event ID: DTMF One 1 (1) 0... .... = End of Event: False .0.. .... = Reserved: False ..00 0111 = Volume: 7 Event Duration: 80 *BLEG SECOND EVENT FROM FREESWITCH* *TO CISCO* No. Time Source Destination Protocol Length Info 5331 23 10.50.47.30 10.50.47.19 RTP EVENT 58 Payload type=RTP Event, DTMF One 1 Frame 5331: 58 bytes on wire (464 bits), 58 bytes captured (464 bits) Ethernet II, Src: Vmware_ab:60:ec (00:0c:29:ab:60:ec), Dst: 64:9e:f3:11:ff:60 (64:9e:f3:11:ff:60) Internet Protocol Version 4, Src: 10.50.47.30 (10.50.47.30), Dst: 10.50.47.19 (10.50.47.19) User Datagram Protocol, Src Port: 24034 (24034), Dst Port: 18768 (18768) Real-Time Transport Protocol [Stream setup by SDP (frame 1113)] [Setup frame: 1113] [Setup Method: SDP] 10.. .... = Version: RFC 1889 Version (2) ..0. .... = Padding: False ...0 .... = Extension: False .... 0000 = Contributing source identifiers count: 0 0... .... = Marker: False Payload type: telephone-event (101) Sequence number: 24717 [Extended sequence number: 90253] Timestamp: 158240 Synchronization Source identifier: 0x57e5d7a1 (1474680737) RFC 2833 RTP Event Event ID: DTMF One 1 (1) 0... .... = End of Event: False .0.. .... = Reserved: False ..00 0111 = Volume: 7 Event Duration: 160

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BLF in Multi-tenant config
by Rick Guyton 29 Apr '12

29 Apr '12

We have 3 different companies in this office and we all share the same freeswitch server, I'll call then CompanyA, CompanyB and CompanyC. CompanyA, B and C all employ their own receptionists, and they are all assigned extension 101 on in their respective domains. However, the receptionists all work very tightly together and basically share work between them. So, I need to assign the BLF status of 101@CompanyA, 101@CompanyB and 101@CompanyC to indicator lights across all phones in the building. My problem is that users in CompanyA can't see the status of 101@CompanyB or 101@CompanyC. Users in CompanyB can't see 101@CompanyA or 101@CompanyC and so on... So, my question is, how do I allow users of one domain in a multi-tenant configuration to see the status of other users in a different domain?

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