March 2014 - FreeSWITCH-users - FreeSWITCH Mailing Lists

ending bridge by request from write function
by Mimiko 26 Mar '14

26 Mar '14

Hello. Lately we have unexpected disconnects. The message in log is: fbe91957-4cec-4928-b5b8-8a7aed579a79 2014-03-21 09:20:45.050899 [DEBUG] switch_ivr_bridge.c:566 sofia/gateway/number ending bridge by request from write function What this message means, and who is faulty: FS or provider? If provider, how I can demonstrate from logs that this is their error? -- Mimiko desu.

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mod_avmd from mod_managed C#
by Malay Thakershi 26 Mar '14

26 Mar '14

Hello, I have been able to integrate mod_avmd and detect beep. As soon as I detect beep, I hang up. I don't want to leave any message if VM is detected. Is it possible? Even though, I hangup and "return" from run function, the code still goes forward. How is that possible? My logic: answer attach event received function (this function checks for avmd_detect variable - if true, it hangs up) start avmd detect whether hung up - if true, return authenticate user by DOB and proceed to survey I use a static variable mBlnIsHungUp and set it to true in hang up hook. I use this flag to detect at various places in "run" method whether call has been hung up. Also, where do I stop avmd? Please help me. Malay Thakershi

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webRTC video
by Holger Esser 26 Mar '14

26 Mar '14

Hi Brian, Thanks for the quick answer. I am not sure if you mean the codec section in the external or internal xml. I was able to make it work in media bypass but I would like to be able to do IVR treatment. I added this to the vars.xml <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G722,PCMU,PCMA,GSM,OPUS,VP8"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU,PCMA,GSM,OPUS,VP8"/> And when I check the registration, I see OUTBOUND-PROXY N/A CODECS IN G722,PCMU,PCMA,GSM,OPUS,VP8 CODECS OUT G722,PCMU,PCMA,GSM,OPUS,VP8 TEL-EVENT 101 DTMF-MODE rfc2833 But still no dice. Sorry for the stupid questions... Thx holger

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Originate fails for freetdm channel
by Kshitij Saxena 25 Mar '14

25 Mar '14

The setup is a Sangoma A101E card being run through Wanpipe 7.0.10 on an Ubuntu 13.10 machine. wanrouter status shows "connected" but "originate freetdm/1/a/xxxxxxxxxx &bridge..." fails with the following error: [ERR] ftdm_io.c:2674 [s1c1][1:1] outgoing_call method not implemented in this span! Note that originate is successful with for other cahnnels (i.e. "originate user/1001 &bridge(9198)" works for example) Would appreciate any guidance. Thanks! Regards, Kshitij Saxena

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webRTC video
by Holger Esser 25 Mar '14

25 Mar '14

Hi, I seem to have a problem with video on webRTC. I use chrome with sipML5. I have to registered clients (user 1 and user 2). I can place a call from one to the other. Audio works fine every time. However, video end to end never works. I checked the logs In the SDP from client 1 I see the VP8 codec [m[m a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time[m[m [m[m a=sendrecv[m[m [m[m a=rtcp-mux[m[m [m[m a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:mQd42qo/WHnzNP97peCV7u4jSbHJ0lDB2CoZNtzb[m[m [m[m a=rtpmap:100 VP8/90000[m[m [m[m a=rtcp-fb:100 ccm fir[m[m [m[m a=rtcp-fb:100 nack[m[m [m[m a=rtcp-fb:100 goog-remb[m[m [m[m a=rtpmap:116 red/90000[m[m However, when freeswitch places the call to client 2, there is no video in the SDP. I thought that the freeswitch core as acting as a B2BUA, but it does not seem to copy the ingress SDP to the egress SDP. I suspect that I am missing something, but I am not sure what. Any help would be appreciated. Thanks

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issue using ignore-early-media variable
by david escartin 25 Mar '14

25 Mar '14

Hello all i'm trying to use the variable ignore-early-media=true to ignore a 183 coming froma provider and sending a 180 instead. but when doing that, (the signaling seems ok as i want), the call has no audio i read on the wiki i had to set bypass_media to false like i was doing transcoding do you have any clue about the particular setup i must use to make it work with audio once the call has been connected? thanks a lot and regards david escartin

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[WARNING] switch_ivr_play_say.c:348 Macro [voicemail_ack]: saved did not match any patterns
by Sean Devoy 25 Mar '14

25 Mar '14

HI All, I think I am a week behind on the latest STABLE git. FreeSWITCH Version 1.2.22+git~20140309T212137Z~65fed130e5~64bit (git 65fed13 2014-03-09 21:21:37Z 64bit) I am getting WARNINGS when users use voicemail, and save a message. In the logfile I see: [WARNING] switch_ivr_play_say.c:348 Macro [voicemail_ack]: saved did not match any patterns A timy bit different on the console, I see: switch_ivr_play_say.c:348 Macro [voicemail_ack]: 'saved' did not match any patterns In /freeswitch/conf/lang/en/vm I have:  <macro name="voicemail_ack"> <input pattern="^(too-small)$"> <match> <action function="play-file" data="voicemail/vm-too-small.wav"/> </match> </input> <input pattern="^(deleted)$"> <match> <action function="play-file" data="voicemail/vm-message.wav"/> <action function="play-file" data="voicemail/vm-$1.wav"/> </match> </input> <input pattern="^(saved)$"> <match> <action function="play-file" data="voicemail/vm-message.wav"/> <action function="play-file" data="voicemail/vm-$1.wav"/> </match> </input> <input pattern="^(emailed)$"> <match> <action function="play-file" data="voicemail/vm-message.wav"/> <action function="play-file" data="voicemail/vm-$1.wav"/> </match> </input> <input pattern="^(marked-urgent)$"> <match> <action function="play-file" data="voicemail/vm-message.wav"/> <action function="play-file" data="voicemail/vm-$1.wav"/> </match> </input> </macro> The file /usr/local/freeswitch/sounds/en/us/callie/voicemail/8000/vm-saved.wav exists. What am I missing? Thanks, Sean

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Handling DTMF using Outbound Socket
by Grant Bagdasarian 25 Mar '14

25 Mar '14

Hello, I'm starting a new application which has the following functionality: A user calls a certain number at which it gets a few option to choose from. The last option allows the user to be transferred to another IVR application (context) inside our FS instance or bridged with a live person (remote party). So far, this won't be hard to setup. Now comes the hard part at which I have no experience in. Upon transferring or bridging the call, FS should create an outbound socket to an external application which will control the call. The external process should monitor the length of the call and once its limit is almost reached, playback a message to the user if it wants to continue the call. The user has to press 1 in order to continue, or nothing or 2 to disconnect. I need to know the following: 1) Is Outbound ESL Sockets the best way to accomplish this? 2) What if the user has been transferred to an IVR application and is currently asked by FS to enter digits, but the monitor asks the user at the same time if it wants to continue the call. How do I handle this? Should I pause the application in which the user currently is in? Is this even possible? 3) Using an external process to control the call, is it possible to playback the call limit reached message to only the caller, and not the bridged to party? 4) Is this whole project even possible? Regards, Grant

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BLF lights sticking
by Tim St. Pierre 25 Mar '14

25 Mar '14

Hello, I just upgraded to 1.2.23 on Friday, from a much older 1.2 version (1.2.11 I think). Since then, I have had problems with BLF lamps not going out when a user hangs up. It's not all the time, just some of the time. I suspect it is when they get a second call, or some other less common situation. All end points are Aastra 67xx, and most are behind NAT. TCP vs. UDP doesn't seem to make a difference. Any ideas as to where to look? Thanks! -Tim

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Week in Review March 16th-22nd
by Kathleen King 25 Mar '14

25 Mar '14

Hello, again. This week in the FreeSWITCH master branch we had an amazing amount of work done with the majority of it going toward build support. The week ended with a total of 159 commits! Some big changes came with Freeswitch attempting to use the system versions of pcre, sqlite, speex, and curl rather than automatically using the in tree version. Also, there have been fixes to be able to run configure specifying the srcdir which allows you to build Freeswitch outside of the source directory. The following bugs were squashed: 8d67246 FS-6374 --resolve mod_rayo sendfax was using uninitialized memory pool Jira: http://jira.freeswitch.org/browse/FS-6374 bd1492e FS-6287 fixed regression from 6e818216e2e615f3241a34253cdea8ee316d9e88 which caused 'operation has no matching challenge' error on outbound invites that are challenged in mod_sofia Jira: http://jira.freeswitch.org/browse/FS-6287 19fc943 FS-6360 Mitigate the CRIME TLS flaw New features that were added: 6d0e684 Upgrade windows build to use pcre 8.34 via getlib and to use speex 1.2rc1 via getlib Jira: http://jira.freeswitch.org/browse/FS-6348 dd242f3 Upgrade windows build to use curl 7.35.0 via getlib Jira: http://jira.freeswitch.org/browse/FS-6346 741cb88 FS-6349 Upgrade windows build to use sqlite 3.8.4.1 via getlib Jira: http://jira.freeswitch.org/browse/FS-6349 714e313 FS-353: add testing hack to use system xmlrpc-c Jira: http://jira.freeswitch.org/browse/FS-353 Improvements in cross platform build supports: 09811b5 FS-6369: fix uninstall on mod_managed Jira: http://jira.freeswitch.org/browse/FS-6369 e85f06e Completely unbundle pcre 9df1727 fix uninitialized variables in mod_translate cba9af4 Drop mod_perl from the debian build temporarily aeebd71 FS-6375 exclude sun from modem support in spandsp Jira: http://jira.freeswitch.org/browse/FS-6375 1470622 Require libcurl as a system dependency 9f7399c FS-6293: fix lua to build using automake, so it works right with srcdir, and avoids the linking to .a is not portable warning Jira: http://jira.freeswitch.org/browse/FS-6293 9a94235 FS-6379: --resolve fix ldns bootstrap/build on solaris Jira: http://jira.freeswitch.org/browse/FS-6379 f6d9027 FS-6375 adjust ifdef for sun to allow for different hep_ipheader struct in libsofia Jira: http://jira.freeswitch.org/browse/FS-6375 3eb7862 FS-6293: fix mod_shout srcdir build Jira: http://jira.freeswitch.org/browse/FS-6293 3b1278a FS-353 Add curl auto PKG_CONFIG_PATH handling for OS X bf611fc Pull mod_v8 out of the default build And many more! In terms of stability these were the use cases that were fixed: f3acb03 FS-6341 fixed deadlock in mod_sofia introduced in commit 340b697 Some packaging updates: fb2c587 Add debian build-deps for mod_opal 6261e5a Drop mod_v8 from debian build for now 90404d5 Purge mod_voipcodecs from debian/control-modules 046df6b Export V=1 for debian builds 54cd0df Support change to system libspeex in debian 83125da update Debian packaging to use configure.ac files Feedback welcome and the referenced commits are in the attached text file with corresponding Jira links. -- Kathleen King Quentus Technologies, INC 1037 NE 65th St Suite 273 Seattle, WA 98115 Main: (877) 211-9337 Cell: (703) 859-3757 kathleen.king(a)quentustech.com

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