August 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

How to write an Application to control freeSWITCH
by Subhash 30 Aug '12

30 Aug '12

Hi All, I am new to freeSWITCH platform. I want to write an application using c++ to control freeSWITCH so please guide me. Thanks in advance. Thanks, Subhash.

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voicemaill
by Kees Varekamp 30 Aug '12

30 Aug '12

I'm trying to get voicemail running on fs1.2 but i keep running into this: 2012-08-30 18:05:27.992966 [ERR] switch_xml.c:2986 Can't find macros tag. Ive got this: <section name="languages" description="Language Management"> ... <X-PRE-PROCESS cmd="include" data="lang/en/*.xml"/> and in lang/en there is en.xml, which has this: <include> <language name="en" say-module="en" sound-prefix="$${sounds_dir}/en/us/callie" tts-engine="cepstral" tts-voice="callie"> <phrases> <macros> <X-PRE-PROCESS cmd="include" data="demo/*.xml"/>   <X-PRE-PROCESS cmd="include" data="vm/sounds.xml"/>  <X-PRE-PROCESS cmd="include" data="dir/sounds.xml"/>  <X-PRE-PROCESS cmd="include" data="ivr/*.xml"/>  </macros> </phrases> </language> </include> Is my config wrong? thanks, Kees

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[freeswitch] <defunct>
by Yuriy Nasida 29 Aug '12

29 Aug '12

Hello guys, What is the best way for understanding the reason why I begin to have [freeswitch] <defunct> process ?I don't use 'exec' directive. CentOS release 6.3 (Final)FreeSWITCH Version 1.1.beta1 (git-de019ab 2012-05-03 15-23-57 +0200) Please advise.Thanks.

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FreeSWITCH Conference Call Today
by Michael Collins 29 Aug '12

29 Aug '12

Hello all, We are having a nice discussion today: http://wiki.freeswitch.org/wiki/FS_weekly_2012_08_22 Mitch Capper and others who have experience will be talking to us about TLS, certs, CA's and things like that. Hopefully we'll all be better prepared to enable TLS transport on our devices and help those who come to the IRC and ML with TLS questions. Talk to you soon! -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org

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sip-router or opensips for good interoperability with freeswitch?
by François Delawarde 29 Aug '12

29 Aug '12

Hi, Which of opensips or sip-router (kamailio) do you guys recommend to use in front of freeswitch to create some big reliable cluster? I'm not sure if there is any huge difference as both seem to share a same origin. Should I just throw a dice?, or is there one that is clearly better than the other (or one I should avoid) in terms of reliability / performance, or friendliness with FS? Thanks, -- François Delawarde <fdelawarde(a)wirelessmundi.com> Wireless Mundi S.L.

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Doubts about T38 passthru
by François Delawarde 29 Aug '12

29 Aug '12

Hi, Whenever my provider puts me on hold, it reINVITES with an SDP containing both audio (sendonly) and image media like so: ... m=audio 5320 RTP/AVP 18 3 8 0 96 c=IN IP4 <ip_address> a=rtpmap:18 G729/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=sendonly a=rtpmap:96 telephone-event/8000 m=image 5322 udptl t38 c=IN IP4 <ip_address> a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxUdpEC:t38UDPRedundancy a=T38FaxRateManagement:transferredTCF Not sure if it's a non-standard SDP, but that's what I get. Now when t38_passthru=true, it only forwards the "image" part to the other leg. When t38_passthru=false, it only forwards the "audio" part. Is there a way to make t38_passthru forward the audio part as well? Or should I just change to a better provider? Thanks, François.

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incoming call hangup if remote SDP contains a=crypto
by Antonio 29 Aug '12

29 Aug '12

Hi all, I'm having a problem to make calls, when the destination answer, the calls is hangup with incompatible destination. Debugging, i could see that is because fs drops the call, because it doesn't support ZRTP. So, is it normal that freeswitch hangs up the call when in the remote party send in the sdp "a=crypto:1 ... " ? Should it ignore and try to established the call using the supported methods in the sdp? Like for video calls, if the remote send video codecs, if we do not have them we don't hangup the call... just make a simple audio call... For incoming calls there is no problem. REMOTE SDP: 2012-08-28 10:37:56.068381 [DEBUG] sofia.c:6122 Remote SDP: v=0 o=- 163879879 1 IN IP4 127.0.0.1 s=webrtc (chrome 22.0.1189.0) - Doubango Telecom (sipML5 r000) t=0 0 a=group:BUNDLE audio video m=audio 57293 RTP/SAVPF 103 104 0 8 106 105 13 126 c=IN IP4 80.28.139.221 a=rtcp:57293 IN IP4 80.28.139.221 a=candidate:3460286729 1 udp 2130714367 192.168.3.1 56196 typ host generation 0 a=candidate:3460286729 2 udp 2130714367 192.168.3.1 56196 typ host generation 0 a=candidate:1004190935 1 udp 2130714367 192.168.10.25 55956 typ host generation 0 a=candidate:1004190935 2 udp 2130714367 192.168.10.25 55956 typ host generation 0 a=candidate:1734362844 1 udp 1912610559 80.28.139.221 57293 typ srflx generation 0 a=candidate:1734362844 2 udp 1912610559 80.28.139.221 57293 typ srflx generation 0 a=candidate:3130178147 1 udp 1912610559 88.9.167.79 59009 typ srflx generation 0 a=candidate:3130178147 2 udp 1912610559 88.9.167.79 59009 typ srflx generation 0 a=candidate:2159818233 1 tcp 1694506751 192.168.3.1 52952 typ host generation 0 a=candidate:2159818233 2 tcp 1694506751 192.168.3.1 52952 typ host generation 0 a=candidate:1968783399 1 tcp 1694506751 192.168.10.25 45540 typ host generation 0 a=candidate:1968783399 2 tcp 1694506751 192.168.10.25 45540 typ host generation 0 a=ice-ufrag:SdZ956Hj4vXnNOhU a=ice-pwd:PQ9aJYFiLGiUfKZqqDXxQdtF a=mid:audio a=rtcp-mux a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:1j6B4cDFNA3Q99XoFujj3wYYex4ThJJNpOgbGifl a=rtpmap:103 ISAC/16000 a=rtpmap:104 ISAC/32000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:106 CN/32000 a=rtpmap:105 CN/16000 a=rtpmap:13 CN/8000 a=rtpmap:126 telephone-event/8000 a=ssrc:2321878499 cname:qlN/YrHlo6j9iIJY a=ssrc:2321878499 mslabel:w4BkX3vCh3pGAdk7f2fqpUd6NT7JS2miWIhu a=ssrc:2321878499 label:w4BkX3vCh3pGAdk7f2fqpUd6NT7JS2miWIhu00 m=video 57293 RTP/SAVPF 100 101 102 c=IN IP4 80.28.139.221 a=rtcp:57293 IN IP4 80.28.139.221 a=candidate:3460286729 1 udp 2130714367 192.168.3.1 56196 typ host generation 0 a=candidate:3460286729 2 udp 2130714367 192.168.3.1 56196 typ host generation 0 a=candidate:1004190935 1 udp 2130714367 192.168.10.25 55956 typ host generation 0 a=candidate:1004190935 2 udp 2130714367 192.168.10.25 55956 typ host generation 0 a=candidate:1734362844 1 udp 1912610559 80.28.139.221 57293 typ srflx generation 0 a=candidate:1734362844 2 udp 1912610559 80.28.139.221 57293 typ srflx generation 0 a=candidate:3130178147 1 udp 1912610559 88.9.167.79 59009 typ srflx generation 0 a=candidate:3130178147 2 udp 1912610559 88.9.167.79 59009 typ srflx generation 0 a=candidate:2159818233 1 tcp 1694506751 192.168.3.1 52952 typ host generation 0 a=candidate:2159818233 2 tcp 1694506751 192.168.3.1 52952 typ host generation 0 a=candidate:1968783399 1 tcp 1694506751 192.168.10.25 45540 typ host generation 0 a=candidate:1968783399 2 tcp 1694506751 192.168.10.25 45540 typ host generation 0 a=ice-ufrag:SdZ956Hj4vXnNOhU a=ice-pwd:PQ9aJYFiLGiUfKZqqDXxQdtF a=mid:video a=rtcp-mux a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:1j6B4cDFNA3Q99XoFujj3wYYex4ThJJNpOgbGifl a=rtpmap:100 VP8/90000 a=rtpmap:101 red/90000 a=rtpmap:102 ulpfec/90000 log when it fails: 2012-08-28 10:38:02.068442 [NOTICE] sofia.c:6853 Channel [sofia/192.168.10.1/sip:2102@192.168.10.30:5062] has been answered 2012-08-28 10:38:02.068442 [DEBUG] sofia_glue.c:3948 Looking for zrtp-hash 2012-08-28 10:38:02.068442 [DEBUG] sofia_glue.c:3926 Deciding whether to pass zrtp-hash between legs 2012-08-28 10:38:02.068442 [DEBUG] sofia_glue.c:3928 CF_ZRTP_PASSTHRU_REQ not set, so not propagating zrtp-hash 2012-08-28 10:38:02.068442 [ERR] sofia_glue.c:4837 a=crypto in RTP/AVP, refer to rfc3711 2012-08-28 10:38:02.068442 [DEBUG] switch_core_session.c:772 Send signal sofia/192.168.1.2_nat/2303(a)test.commsmundi.com [BREAK] 2012-08-28 10:38:02.068442 [DEBUG] switch_channel.c:2924 (sofia/192.168.1.2_nat/2303(a)test.commsmundi.com) Callstate Change RINGING -> HANGUP 2012-08-28 10:38:02.068442 [NOTICE] switch_channel.c:3412 Hangup sofia/192.168.1.2_nat/2303(a)test.commsmundi.com [CS_EXECUTE] [INCOMPATIBLE_DESTINATION] -- Best regards, _________________________ António Silva E-mail:asilva@wirelessmundi.com

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Setup Voicemail to Expire after X amount of Days.
by Chris Ferreira 28 Aug '12

28 Aug '12

How can I set FreeSWITCH up so that OLD voicemails get deleted automatically? I havent found anything in the Wiki or other documentation. Your input is much appreciated. Thanks, -Chris

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FreeSWITCH Weekly News and Notes
by Michael Collins 28 Aug '12

28 Aug '12

Hello all! Apologies for the delay on this week's news and notes. Yesterday was quite busy, but things are going well. I am happy to report that we have submitted the first chapter of the new FreeSWITCH book to the publisher! We've also conferred with a few members of the community and convinced Packt<http://www.packtpub.com>to let us add some bonus content! Stay tuned for more previews and teases. Right now it's still early in the game so I don't want to reveal too much. On last week's conference call<http://wiki.freeswitch.org/wiki/FS_weekly_2012_08_22>we discussed a number of things. First, we did a follow up to Ken's previous discussion <http://wiki.freeswitch.org/wiki/FS_weekly_2012_08_15>about the stable 1.2 branch and using git. Second, we talked a bit about Vestec <http://www.vestec.com> and the great ASR application contest. (More information is forthcoming!) Lastly, we had Mitch Capper discussing the latest version of the FSClient <http://wiki.freeswitch.org/wiki/FSClient>Windows softphone. If you haven't tried it out I highly recommend it. It's now stable and feature-enabled to the point that I've discontinued using X-Lite or Jitsi. Tomorrow we hope to have a discussion about TLS<http://wiki.freeswitch.org/wiki/Tls>. We have several community members who are experienced with key and certificate management and we will be calling upon them to share their experience with the rest of us. After that we will have an open discussion. Cheers! -- Michael S Collins ClueCon Team http://www.cluecon.com 877-7-4ACLUE

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voicemail to email not working
by Abaci 28 Aug '12

28 Aug '12

I'm trying to get FreeSWITCH to send my voicemail to my email and can't it to work. I have sSMTP installed and working (tested using voicemail.tpl) I see in the FreeSWITCH logs *"switch_utils.c:761 Emailed file [/tmp/mail.13461330724a96] to [me(a)mydomain.com]" *however in /var/log/maillog I see nothing and I don't get the email. does someone have a clue on what might be wrong here, or is there any debug I can enable for this? _*in switch.conf.xml I have:*_ <param name="mailer-app" value="sendmail"/> <param name="mailer-app-args" value="-t"/> _*in voicemail.conf.xml I have:*_ <email> <param name="template-file" value="voicemail.tpl"/> <param name="notify-template-file" value="notify-voicemail.tpl"/>  <param name="date-fmt" value="%A, %B %d %Y, %I %M %p"/>  </email> _*in the directory (xml_curl) I have:*_ <document type="freeswitch/xml"> <section name="directory"> <domain name="mydomain.com"> <user id="200"> <params> <param name="password" value="cj987r5y95"/> <param name="vm-password" value="1234"/> <param name="vm-enabled" value="true"/> <param name="vm-email-all-messages" value="true"/> <param name="vm-mailto" value="me(a)mydomain.com"/> <param name="vm-mailfrom" value="Voicemail Server <voicemail(a)mydomain.com>"/> <param name="vm-attach-file" value="true" /> <param name="dial-string" value="${sofia_contact(200(a)mydomain.com)}"/> </params> <variables> <variable name="toll_allow" value="domestic,international,local"/> <variable name="user_context" value="routing"/> <variable name="effective_caller_id_name" value="test user"/> <variable name="effective_caller_id_number" value="2125551212"/> <variable name="outbound_caller_id_number" value="2125551212"/> </variables> </user> </domain> </section> </document>

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