September 2014 - FreeSWITCH-users - FreeSWITCH Mailing Lists

WebRTC meeting Norfolk, 15 October 2014
by Daniel Pocock 10 Sep '14

10 Sep '14

I'll be in Norfolk, VA for xTupleCon in October On 15 October, there will be two events for WebRTC: 14:15 a talk about the xTuple WebRTC extension at xTupleCon - must register for xTupleCon to attend this 17:30 a technical / developer workshop at xTuple's offices - free, anybody welcome, even if not attending xTupleCon, RSVP through Eventbrite[1] Please see my blog[2] for more comments about all of this and feel free to email me in advance if you have questions about it or if you may like to meet up there. 1. http://www.eventbrite.com/e/browser-based-webrtc-telephony-for-web-apps-wor… 2. http://danielpocock.com/xtuplecon-webrtc-talk-schedule-change

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Channel is hungup and application 'curl' does not have the zombie_exec flag.
by Phillip Jones 10 Sep '14

10 Sep '14

Hi all, This a very simple custom VM that plays a greeting, bong and then records the callers message. After the caller has hung up, I need to call a webservice through curl. The zombie_exec flag gets me to the last line but the curl does not execute complaining: "Channel is hungup and application 'curl' does not have the zombie_exec flag." How do I apply the zombie flag to CURL? <condition field="${ForwardingNumber}" expression="([1-9]),(.*)" break="never"> <action application="set_zombie_exec"/> <action application="answer" /> <action application="sleep" data="2000"/> <action application="playback" data="$${user_vm_greeting}\$2"/> <action application="playback" data="tone_stream://$${vm_bong}"/> <action application="record" data="$${user_vm_message}/${accountno}-${uuid}.wav $${maxreclen} $${silencethreshold} $${silencehits}"/> <action application="curl" data=" http://localhost:58784/api/Voicemail/InsertVoicemail"/> </condition> Thanks in anticipation. Phil

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NOTIFY's Cseq too high?
by Yehavi Bourvine 10 Sep '14

10 Sep '14

Hi, My Polycom phones stopped showing presence today. They reply with "loop detected" after they receive the NOTIFY message. In their Syslog they complain that Cseq is outside boundaries... The Cseq for INVITE is much smaller. How can I reset the Cseq to lower values? Thanks! __Yehavi:

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Double NAT
by k 10 Sep '14

10 Sep '14

Hi, all ! I'm looking for update to the problem mentioned here : https://wiki.freeswitch.org/wiki/Auto_NAT ----------------- Doublenat with eyebeam/snom : Eyebeam/snom1 <> NAT <> Internet <> NAT <> FS Also Eyebeam/snom2 <> Internet <> NAT <> FS Both "work" with audio both ways. snom1 could call snom2 and vice versa and also other phones on the LAN. Found that call is hung up with a "BYE" from FS stating reason is "ACK timeout" although "ACK" was being received at FS (and there was still audio, so shouldn't be due to RTP timeouts). Not sure if this is due to some wrong setup etc. Will update once it is clear. Eyebeam had stun enabled. FS had stun disabled and autonat enabled. Router was netgear FVS318 wireless. Same issue with snom 320 as well. ----------------- I'm experiencing exact the same thing with the following scheme : eyebeam <> NAT <> FS <> NAT <> eyebeam stun and ice are enabled on eyebeam, FS is configured for inbound_bypass_media, TLS, SRTP works well until both clients behind nat. Experience random hangups in several seconds .. several minutes

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Params to FS Native.Api() changed. How to get handle of CoreSession?
by Srini K 10 Sep '14

10 Sep '14

Hi, In older version of FreeSWITCH I have been using FreeSWITCH.Native.Api(). For eg, string uuid = new FreeSWITCH.Native.Api().ExecuteString("create_uuid"); activeCalls = new FreeSWITCH.Native.Api().ExecuteString("show calls") I updated FS to latest version. Now Native.Api expects CoreSession as param. FreeSWITCH.Native.Api(FreeSWITCH.Native.CoreSession) How to get handle of CoreSession. Iam using mod_managed. Thanks in advance. Srini

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user directory is ignoring user_context
by ik 10 Sep '14

10 Sep '14

Hello, I have a weird issue with freeswitch (Version 1.4.8 git 1fe89f5 2014-08-21 18:57:58Z 64bit) sometimes. It also happens to me on Freeswitch 1.2. It ignores the extension's user_context, defined like so: <variable name="user_context" value="testing" /> <include> <user id="3002"> <params> <param name="password" value="******" /> </params> <variables> <variable name="accountcode" value="3002" />  <variable name="user_context" value="testing" /> <variable name="effective_caller_id_name" value="Desktop machine" /> <variable name="effective_caller_id_number" value="3002" /> </variables> </user> </include> When I do list_users, i can see it: 3002|testing|XXX.XXX.XXX.XXX|default|error/user_not_registered||Desktop machine|3002 (and when it is registered I can see it registered) And when the user is registered I see it also: 2014-09-08 18:05:25.397965 [DEBUG] sofia_reg.c:3013 event_add_header -> 'record_stereo' = 'true' 2014-09-08 18:05:25.397965 [DEBUG] sofia_reg.c:3013 event_add_header -> 'default_gateway' = 'example.com' 2014-09-08 18:05:25.397965 [DEBUG] sofia_reg.c:3013 event_add_header -> 'default_areacode' = '918' 2014-09-08 18:05:25.397965 [DEBUG] sofia_reg.c:3013 event_add_header -> 'transfer_fallback_extension' = 'operator' 2014-09-08 18:05:25.397965 [DEBUG] sofia_reg.c:3013 event_add_header -> 'accountcode' = '3002' 2014-09-08 18:05:25.397965 [DEBUG] sofia_reg.c:3013 event_add_header -> 'user_context' = 'testing' 2014-09-08 18:05:25.397965 [DEBUG] sofia_reg.c:3013 event_add_header -> 'effective_caller_id_name' = 'Desktop machine' 2014-09-08 18:05:25.397965 [DEBUG] sofia_reg.c:3013 event_add_header -> 'effective_caller_id_number' = '3002' However, when I dial the extension, it enter a different context: 2014-09-08 18:05:38.897955 [INFO] mod_dialplan_xml.c:558 Processing 3001 <3002>->3000 in context external_dialer xml_locate dialplan <context name="testing"> <extension name="shout"> <condition field="destination_number" expression="^(300\d)$"> <action application="answer"></action> <action application="playback" data="shout:// 208.53.164.178:80"></action> </condition> </extension> </context> It usually happens after a while, when first it enters the proper context, and later it starts to go somewhere else. even on ./freeswitch -stop and ./freeswitch -ncwait it goes to the wrong context. Am I doing something wrong, or missing something ? Thanks, Ido

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effective_caller_id_name in user directory not being passed to SIP endpoint
by Gavin Hamill 10 Sep '14

10 Sep '14

Hi all, I'm starting out with FS 1.4.7 on CentOS 6 (installed via the FS repo at http://files.freeswitch.org) and have a simple working config with two sofia profiles, internal and external. I can register SIP clients, call between internal extensions and call external numbers through a SIP gateway. Each user in directory/default/*.xml looks like this: <include> <user id="106"> <params> <param name="password" value="xxx"/> </params> <variables> <variable name="accountcode" value="106"/> <variable name="user_context" value="default"/> <variable name="effective_caller_id_name" value="Gavin Hamill"/> <variable name="outbound_caller_id_name" value="Gavin Hamill"/> </variables> </user> </include> However, when I call from 106 -> 105, I only see '105' appear in the destination client when I would expect to see 'Gavin Hamill' If I force the effective_caller_id_name to be a fixed string: <context name="default"> <extension name="bashton_office_staff"> <condition field="destination_number" expression="^(10[0-9])quot;> <action application="set" data="effective_caller_id_name=QWERTYUIOP" /> <action application="bridge" data="user/$1"/> </condition> ... </context> Then sure enough calls always appear as coming from 'QWERTYUIOP' At https://confluence.freeswitch.org/display/FREESWITCH/XML+User+Directory there is a section 'Variables' which states: Any variables defined in the domain or user will be defined as channel variables when there is a call to user or when there is an inbound calls from that user. That's exactly the behaviour I was expecting - any pointers where I'm going wrong would be warmly received. Cheers, Gavin

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Freeswitch Week in Review(Master Branch) August 31st-September 6th
by Kathleen King 09 Sep '14

09 Sep '14

Hello, again. This week in the FreeSWITCH master branch we had 48 commits. There was quite a bit of work done on mod_verto. A few of the features that were committed this week are: HTTP utils were added to the core, added http query string parsing and improve http header parsing to core, added initial support of http handing in mod_verto, added a param to let the socket stay open even if not websockt, and added a patch to throw an event with the gateway pings for better monitoring of gateway state changes in mod_sofia. The following bugs were squashed: 1b612fe FS-6757 FS-6713 Fix for choppy audio recording Jira: https://jira.freeswitch.org/browse/FS-6757 7d11e19 FS-6783 FS-6775 Check for null hash or private while holding lock in mod_hash 47b7c44 Fix a problem where radius channel variables were missing in mod_xml_radius 9d9feb6 Make the tzfile header check look for "TZif" instead of "TZif2" 1a4e6e3 FS-6796 #comment avoid use c++ string since it causing troubles Jira: https://jira.freeswitch.org/browse/FS-6796 59cd3c7 FS-6796 #resolve hack mod_lua to read correct binary string Jira: https://jira.freeswitch.org/browse/FS-6796 4e07845 Fix incorrect string termination in mod_verto 023f6e1 Fix http header parsing, now it accepts short header in mod_verto New features that were added: ce96d95 Add support for cache directory make configs use specific directory variables instead of just base_dir so that one configuration can be used for fhs and legacy installations 5c15e85 Added SWITCH_CURRENT_APPLICATION_RESPONSE_VARIABLE on application errors mod_spandsp 6146c5c FS-3867 Add special case for oracle alive query in core Jira: https://jira.freeswitch.org/browse/FS-3867 f0b9407 Add http utils to core 31f78d8 Add the ability to read from a stream 602e82d Add param to let the socket stay open even if not websockt, thanks Anthony. b255484 Added initial support of http handing in mod_verto 93bd583 Add uptime property to mod_sofia gateways 5c29d8d Show gateway uptime in seconds 56e4afb FS-6595: Enable pause_play and file_seek for conference member fnode #resolve Jira: https://jira.freeswitch.org/browse/FS-6595 57bd4eb FS-5992 Add `valet_park ... auto out ...` to syntax string 388e963 Added patch to throw an event with the gateway pings for better monitoring of gateway state changes in mod_sofia 9218775 Update openssl 7f8cc54 Add basic http virtual host support and fix some leaks in mod_verto a9b9155 Add HTTP Basic auth to mod_verto f5c0c13 Add http query string parsing and improve http header parsing to core f361655 Add ability to parse x-www-form-urlencoded post body in mod_verto 7be6047 Respond to OPTIONS and only allows GET and HEAD on static resources in mod_verto c02b242 Refactor http parsing and prevent read body more than content-length a845755 HTTP 1.1 keepalive support in mod_verto Improvements in cross platform build supports: 5fadb52 Fix build on gcc-4.9 5dfea66 Build fix for mod_xml_radius 76b093a Fix trivial windows compiler warning for vs2010 In terms of stability these were the use cases that were fixed: 8c0a1d1 FS-6783 FS-6775 Reintroduce memory leak in mod_hash fixed in commit 3e24b3e2c2b28cc672943e75b009377d33f1b01b to avoid crash 82a28e7 FS-6783 FS-6775 Fixing the leak reintroduced in commit 8c0a1d17e241622f25b9bd7cc68c575b7892adda in mod_hash 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: (253) 617-3210 Cell: (703) 859-3757 kathleen.king(a)quentustech.com

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different transport for outbound proxy and destination
by Yuriy Nasida 09 Sep '14

09 Sep '14

Hi there, I would like to have this scenario: FS ----UDP----> Opensips ---- TLS --- endpoint I suppose that I have to have ;transport=tls in INVITE from FS to opensips but when I do that i getting sofia_glue.c:2326 TLS not supported by profile Can I send INVITE by UDP but let opensips know that it should USE TLS for endpoint ? Please advice Thanks.

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record_post_process_exec_app not fireing
by Phillip Jones 09 Sep '14

09 Sep '14

Anyone had any luck with this setting. I just can not the app to run. <condition field="${ForwardingNumber}" expression="([1-9]),(.*)" break="never"> <action application="answer" /> <action application="sleep" data="2000"/> <action application="playback" data="$${user_vm_greeting}\$2"/> <action application="playback" data="tone_stream://$${vm_bong}"/> <action application="set" data="record_post_process_exec_app=vm_test.lua" /> <action application="record" data="$${user_vm_message}/${accountno}-${uuid}.wav $${maxreclen} $${silencethreshold} $${silencehits}"/> </condition> vm_test.lua: freeswitch.consoleLog("INFO", "Hello!!\n") freeswitch.consoleLog("INFO", "Hello!!\n") On running the dialplan I see: 2014-09-09 10:51:29.964998 [DEBUG] mod_dptools.c:1435 sofia/internal/ 1000(a)192.168.1.22 SET [record_post_process_exec_app]=[vm_test.lua] Everything works - the file is recorded. On leg A hangup up test.lua is not run. Any thoughts? FreeSWITCH Version 1.5.12b~64bit Windows Thanks PJ

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