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FreeSWITCH Week in Review(Master Branch) April 20th-26th
by Kathleen King 29 Apr '14

29 Apr '14

Hello, again. This week in the FreeSWITCH master branch we had 17 commits. The range of work done last week touched mod_sofia, mod_xml_curl, mod_directory, and ESL. Thing seem to be stabilizing on the master branch after March's 100+ commits per week. Reminder: Sign up for ClueCon before May 1st(this Thursday) using the code CLUECON2013 and you can get a discount of $100! Also, if you have a presentation proposal or know of someone that does, please submit them to marketing(a)cluecon.com <mailto:marketing@cluecon.com>. The following bugs were squashed: d171e15 fix issues with heartbeat in bypass media mode 61e22e8 fixed a bug where websocket secure was closing 1 second after it was opened in libsofia Jira: http://jira.freeswitch.org/browse/FS-6476 7e15c93 FS-6482 fixed bug with contact header showing the incorrect port in mod_sofia Jira: http://jira.freeswitch.org/browse/FS-6482 New features that were added: 68d863a FS-6470 added patch to report curl errors for easier debugging in mod_xml_curl Jira: http://jira.freeswitch.org/browse/FS-6470 1aa2fc7 FS-3332 add a parameter to be able to specify a group name to include in lookup list in mod_directory Jira: http://jira.freeswitch.org/browse/FS-3332 0ba59aa Expanding on 052638186b2784c4142089008adf4289d40c80bc to support boolean tel url params in mod_sofia ba6b9c5 ESL-91 increased buffer size of arg_buf from 512 to 4096 and send_buf from 1292 to 5120 to allow for larger commands in ESL In terms of stability these were the use cases that were fixed: bcc2aa9 Fixing memory leak that would occur when reloading mod_xml_curl 2e45fc0 fix memory leaks in mod_h323. Thanks to Sharath from packetizer forum. 96c6abb Fix buffer overrun that could have been exploited with a crafted skinny packet designed to exploit this mistake in bounds checking And some miscellaneous work: 25c9b61 preserve old caller ID behavior if display name is not given in mod_rayo 1eab82b relaxed the recent caller ID restrictions in mod_rayo 2e0cd74 FS-6477 if caller is on hold and can't speak then label them as inactive in mod_conference Jira: http://jira.freeswitch.org/browse/FS-6477 f27fe7e BNPH-6470 follow up to commit 68d863a179c90d2524f8e761dedb3aebd09c3e75 removing the original line that performed the curl request to avoid a double request in mod_xml_curl 4def48b FS-6476 partially reverted commit 61e22e8b500afd8b03f32441398c8cebcf7cbb11 1affd78 we only call this with values, and it would crash if you passed it null anyways, just assert to make it 100% clear in mod_sofia abd7882 contact->m_url is an array, don\'t check array for zero as its useless Feedback is welcome at kathleen.king(a)quentustech.com and the commits for this week are in the attached file along with the 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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Bypass media mode with "intercept" application
by FSX . 29 Apr '14

29 Apr '14

I'm trying to use GV callback feature in FS and having trouble with making it work in bypass media mode. Incoming calls are working well. Call comes to public context, then it's transferred to local domain context. Then it goes via: <action application="set" data="bypass_media=true"/><action application="bridge" data="..."/>RTP streams directly between remote SIP (caller party) and local SIP client, as it should be by design (calls have better audio quality and don't use resources of local SIP server). Now, I try to do the same when I use GV callback feature. Local dial plan (as recommended in Google Voice API) sets hash with current "uuid" and runs system call to make callback. Then in public context dial plan incoming call is accepted, using "intercept" application with saved "uuid". Exactly as it's recommended. It's working well, but in this case I can't get bypass mode whatever I try to do. * I tried to set "bypass_media=true" before run system call to make callback * I tried to set "bypass_media_after_bridge=true=true" before run system call to make callback * I tried to export those variables from leg A to incoming call (leg B) * I tried to set it before executing "intercept" application on incoming call * I tried to apply global setting to make sure that FS is always using bypass mode ("inbound-proxy-media" set to "true" in "sip_profiles\internal.xml") Nothing works so far... FS always ignores those settings and proxies RTP stream. How to use bypass media mode in this scenario? Thank you in advance

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Why is loopback required to originate from endpoints registered on multiple profiles?
by Alex Crow 29 Apr '14

29 Apr '14

Hi, Is it part of the implementation of sofia that means, even with a shared reg and presence db, that if I have the same extension registered on both the default profile and a NAT profile, that originate user/1000 <destno> seems to only ring one of the registered endpoints (and it seems so far random which one rings), while originate loopback/1000 <destno> and for an existing call bridge user/1000 happily rings both endpoints? Cheers Alex

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OS X Wiki Updates March 2014
by Mario G 29 Apr '14

29 Apr '14

Posting here for interested parties and in case someone searches for OS X here. I finally completed the long over due 10.9 (I waited for Xcode 5.1) as well as updated all the installation pages. I intended to update the main page a lot but it has to wait for now. The install updates were overdue, many changes (some listed below) to make them better. As usual, I tested everything I wrote. Yes… I actually now have 8 different partitions for testing, 10.6 through 10.9 times two (one for homebrew one for alternate script testing). That is why I don't do macports, I would need 4 more partitions since I have to test them all! Homebrew has been perfect for FreeSwitch. I have been working since Dec to develop these changes and testing. Hope they help you! Mario G Changes to the Main OS X Page http://wiki.freeswitch.org/wiki/Installation_and_Setup_on_OS_X * Updated test dates as usual * Added 10.9 link * Added info about test systems (fresh installs) Install Pages OS X 10.9 installation of prerequisites and FreeSwitch http://wiki.freeswitch.org/wiki/Installation_on_OS_X_10.9_Mavericks * New page. * New procedure to obtain Command Line Tools. OS X 10.8 installation of prerequisites and FreeSwitch http://wiki.freeswitch.org/wiki/Installation_on_OS_X_10.8_Mountain_Lion * Changes to bring info up to date. * Many changes to enhance readability and understanding. * Xcode now points directly to Xcode in app store for download. * Homebrew and alternate install clarification. OS X 10.7 installation of prerequisites and FreeSwitch http://wiki.freeswitch.org/wiki/Installation_on_OS_X_10.7_Lion * Changes to bring info up to date. * Many changes to enhance readability and understanding. * Xcode 4.6 changes since removal from app store. * Homebrew and alternate install clarification. * Warning about mod_v8 removal. OS X 10.6 installation of prerequisites and FreeSwitch http://wiki.freeswitch.org/wiki/Installation_on_OS_X_10.6_Snow_Leopard * Changes to bring info up to date. * Xcode removal from installation procedure. * Homebrew only install. * Warning about mod_v8 removal. * Previous changes: Now uses Homebrew, Apple removing Command Line Tools for 10.6. * Moved 10.6 prerequisite instructions from alternate install page to here, OS X Installation Alternatives illustrates how to install without Homebrew: http://wiki.freeswitch.org/wiki/Installation_on_OS_X_Alternatives * Major changes/rewrite * Added install script * Added warnings * Added requirements list. * Added makefile option (only thing I do not test) * Remove all 10.6 references * Changed macports comment Old/Deleted Page http://wiki.freeswitch.org/wiki/Installation_and_Tips_for_Mac_OS_X * Comments with link to new page (tried to do timer redirect but I don't know how on the wiki)

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Weekly News and Notes
by Ken Rice 29 Apr '14

29 Apr '14

Hey FreeSWITCHers Join us tomorrow April 30th, 1PM Eastern on ClueCon Weekly. This week¹s topic, Tracking Down troubled commits in the source tree with git bisect. Where sip:888@conference.freeswitch.org or +1 919-386-9900 Whats git bisect you ask? Ever had one of those issues where you know it worked in a previous version, but now its broken? Git bisect helps you quickly isolate which commit changed things on you. Join us tomorrow as Travis Cross breaks out a little tutorial on how to utilize bisect! Have you registered for ClueCon? If not, get those registrations in today and tomorrow, use the discount code CLUECON2013 and get last years price! This code is only valid through the end of April and time is rapidly running out! Do you have a presentation you would like to show off at ClueCon? Be sure to submit your proposal to marketing(a)cluecon.com before its too late! Speaking slots are still available! -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH

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Re: [Freeswitch-users] Consulting help
by andretodd＠verizon.net 29 Apr '14

29 Apr '14

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Re: [Freeswitch-users] Consulting help
by andretodd＠verizon.net 29 Apr '14

29 Apr '14

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hangup with cause:WRONG_CALL_STATE
by rockxsj 29 Apr '14

29 Apr '14

Hi guys! I found that the some of the incoming call will hangup with case: WRONG_CALL_STATE,but else will be ok.After a hour,almost every incoming call will be hangup with the same cause.Then I will restart the freeswitch,it will server like before:some call will be hangup with the cause... 28e10f2e-c851-11e3-a146-85a8fadad1b0 2014-04-20 14:01:15.498169 [WARNING] switch_core_state_machine.c:567 28e10f2e-c851-11e3-a146-85a8fadad1b0 sofia/internal/123456(a)sipsj.236.cn Abandoned 28e10f2e-c851-11e3-a146-85a8fadad1b0 2014-04-20 14:01:15.498169 [NOTICE] switch_core_state_machine.c:570 Hangup sofia/internal/123456(a)sipsj.236.cn [CS_NEW] [WRONG_CALL_STATE] What does Wrong call state mean and what can I do to fix this issue? Everything else is working. I can make outgoing calls and call other extensions. Thanks in advance.

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Yealink T48G and TLS
by Iskren Hadzhinedev 29 Apr '14

29 Apr '14

Hello everyone, Just got a couple of new Yealink T48G phones and I am having a couple of rather weird (at least for me) issues with them. Whenever I enable TLS authentication on the phones, they register with the FreeSWITCH box but there's no media on the outbound calls and I'm unable to get any incoming calls at all. If I switch the authentication protocol to TCP or UDP it's all working good. I tried enabling/disabling SRTP for all 3 protocols and it kept the behaviour consistent. All calls made are local (registered to the same FreeSWITCH box) and only these phones have any issues with calls. I can call someone from the Yealink and then add another person in a 3-way conference. That way the two remote parties can hear eachother, but the Yealink is dead silent. Here's the profile on which all phones are connected: <profile name="local"> <domains> <domain name="all" alias="true" parse="false"/> </domains> <settings> <param name="debug" value="0"/> <param name="sip-trace" value="no"/> <param name="sip-capture" value="no"/> <param name="watchdog-enabled" value="no"/> <param name="watchdog-step-timeout" value="30000"/> <param name="watchdog-event-timeout" value="30000"/> <param name="log-auth-failures" value="true"/> <param name="forward-unsolicited-mwi-notify" value="false"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="liberal-dtmf" value="true"/> <param name="dtmf-duration" value="2000"/> <param name="inbound-codec-prefs" value="SILK,OPUS,G722,PCMU,PCMA,GSM"/> <param name="outbound-codec-prefs" value="PCMU,PCMA,GSM"/> <param name="rtp-timer-name" value="soft"/> <param name="rtp-ip" value="$${local_ip_v4}"/> <param name="sip-ip" value="$${local_ip_v4}"/> <param name="hold-music" value="local_stream://moh"/> <param name="record-path" value="$${base_dir}/recordings"/> <param name="record-template" value="${caller_id_number}.${target_domain}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/> <param name="manage-presence" value="true"/> <param name="inbound-codec-negotiation" value="generous"/> <param name="tls" value="true"/> <param name="tls-only" value="false"/> <param name="tls-version" value="tlsv1"/> <param name="tls-bind-params" value="transport=tls"/> <param name="tls-sip-port" value="5061"/> <param name="tls-cert-dir" value="$${base_dir}/conf/ssl"/> <param name="tls-verify-date" value="true"/> <param name="inbound-late-negotiation" value="true"/> <param name="inbound-zrtp-passthru" value="true"/> <param name="nonce-ttl" value="60"/> <param name="auth-calls" value="yes"/> <param name="inbound-reg-force-matching-username" value="true"/> <param name="auth-all-packets" value="false"/> <param name="ext-rtp-ip" value="$${local_ip_v4}"/> <param name="ext-sip-ip" value="$${local_ip_v4}"/> <param name="challenge-realm" value="auto_from"/> </settings> </profile> and (due to their sizes) a tport log, a siptrace for an outgoing call from the Yealink and an incoming call (that never rings the phone) with TLS enabled. The whole setup is: FreeSWITCH -- Internet -- NAT Router -- Yealink and Android phone (in different subnets so no direct LAN communication between them) Any thoughts are greatly appreciated. Thanks in advance! Kind regards,-- Iskren Hadzhinedev

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Bypass media mode with "intercept" application
by FSX 28 Apr '14

28 Apr '14

I'm trying to use GV callback feature in FS and having trouble with making it work in bypass media mode. Incoming calls are working well. Call comes to public context, then it's transferred to local domain context. Then it goes via: <action application="set" data="bypass_media=true"/> <action application="bridge" data="..."/> RTP streams directly between remote SIP (caller party) and local SIP client, as it should be by design (calls have better audio quality and don't use recourses of local SIP server). Now, I try to do the same when I use GV callback feature. Local dial plan (as recommended in Google Voice API <https://wiki.freeswitch.org/wiki/Google_Voice_API> ) sets hash with current "uuid" and runs system call to make callback. Then in public context dial plan incoming call is accepted, using "intercept" application with saved "uuid". Exactly as it's recommended. It's working well, but in this case I can't get bypass mode whatever I try to do. * I tried to set "bypass_media=true" before run system call to make callback * I tried to set "bypass_media_after_bridge=true=true" before run system call to make callback * I tried to export those variables from leg A to incoming call (leg B) * I tried to set it before executing "intercept" application on incoming call * I tried to apply global setting to make sure that FS is always using bypass mode ("inbound-proxy-media" set to "true" in "sip_profiles\internal.xml") Nothing works so far... FS always ignores those settings and proxies RTP stream. How to use bypass media mode in this scenario? Thank you in advance -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Bypass-media-mode-with-interc… Sent from the freeswitch-users mailing list archive at Nabble.com.

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