January 2015 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Error whith xml_curl
by Antony Aguirre Morales 27 Jan '15

27 Jan '15

I set my freeswitch with the directory database, now by registering my extensions are connected correctly, but when making a call of the extension extention 1 to 2 me throws the following error "01/27/2015 10: 22: 35.158536 [ERR] mod_dptools.c: 3977 No dial-string available, Please check your user directory." and directly sends the voice mailbox of the extension. Thanks!

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Bug Marshalls Needed!
by Brian West 27 Jan '15

27 Jan '15

FreeSWITCHers, We're currently in need of bug marshalls, The job is basically to triage JIRA's make sure the data is correct, try to replicate the issue if the description is clear enough or ask for more info about the issue. If you're interested in helping out please email krice(a)freeswitch.org or myself. Thanks, -- *Brian West* brian(a)freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest

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lua + aes|md5
by Денис 27 Jan '15

27 Jan '15

Hello. I need module md5 or aes for checksums, but modules from luarocks not work. Who can make luarocks version for freeswitch with 5.2 version? Or system RPM with 5.2 My OS is CentOS 6.

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how to transfer all conference members to an extension on stop-talking event.
by Aqs Younas 27 Jan '15

27 Jan '15

Hi, list. I need to transfer all conference members to some other extension whenever there is any stop-talking event. But don't know how can i do so? I am using python script as inbound connection to listening to events. Whenever there is stop-talking event I need to transfer all conference members to another dialplan extension. Your help would be much appreciated. Thanks.

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Getting Redirecting number in Events (ISDN)
by DP Siddharth 27 Jan '15

27 Jan '15

Hi, I am using ISDN (PRI) for handling calls. In a use case we need Redirecting number as part of freeswitch events. I verified various events but we get only caller/called. I was looking into the code & found for SS7 this is available but not for ISDN, just need confirmation if any way possible to get this. I am using 1.2.24 (1.2.stable) freeswitch for testing. following is ISDN SETUP message for reference: Prot Disc:Q.931/I.451 (0x08) Call Ref:242E (Originating side) Type:SETUP (0x5) Sending complete: Bearer Capability:Coding:ITU-T(0) TransferCap:Speech(0) TransferRate:64 Kbit/s(16) L1Prot:G.711 A-Law(3) Channel Id:No:13 Type:B-chans(3) Preferred/Implicit Calling Party Number:8860128000(l:10) plan:isdn(1) type:national(2)scr:network, provided(3) pres:allowed(0) Called Party Number:01725218700(l:11) plan:isdn(1) type:subscriber(4) Redirecting Number:53703417(l:8) plan:isdn(1) type:national(2)scr:user, not screened(0) pres:restricted(1)reason:Call forwarding unconditional(15) Redirecting Number:53703417(l:8) plan:isdn(1) type:national(2)scr:user, not screened(0) pres:restricted(1)reason:Call forwarding unconditional(15) [ 08 02 24 2e 05 a1 04 03 80 90 a3 18 03 a1 83 8d 6c 0c 21 83 38 38 36 30 31 32 38 30 30 30 70 0c c1 30 31 37 32 35 32 31 38 37 30 30 74 0d 21 20 8f 39 36 35 33 37 30 33 34 31 37 74 0d 21 20 8f 39 36 35 33 37 30 33 34 31 37 ] -- Thanks & Regards, D P Siddharth Director (Platform) Knowlarity Communications Ph: +919999115231 dp.siddharth(a)eng.knowlarity.com <emailID(a)knowlarity.com> *"Come together to build a lasting world-class cloud telephony company that helps businesses grow"*

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Freeswitch don't send keepalive
by Tanguy 27 Jan '15

27 Jan '15

Hello I'm a asterisk user, and i would like to try freeswitch for having a real multi tenant system. I configured FS with fusionpbx. My FS is installed on a VPS ( has a public IP ), my extensions are behind a nat router. After few minutes my phone became unreachable for inbound call. After a outbound call, the phone is again reachable for few minutes. I captured the traffic on server side with tcpdump, unlike my asterisk server, freeswitch never send to the phone a keepalive packet ( SIP OPTIONS ) I added theses options in sip_profiles/internal.xml <param name="aggressive-nat-detection" value="true"/> <param name="enable-timer" value="false"/> <param name="NDLB-received-in-nat-reg-contact" value="true"/> <param name="NDLB-force-rport" value="true"/> <param name="NDLB-broken-auth-hash" value="true"/> <param name="sip-force-expires" value="240"/> <param name="all-reg-options-ping" value="true"/> <param name="registration-thread-frequency" value="30"/> <param name="unregister-on-options-fail" value="true"/> <param name="nat-options-ping" value="true"/> Please note the strange expiry delay. After being powered off, the phone stays registered. freeswitch@internal> sofia status profile internal reg Registrations: ================================================================================================= Call-ID: 24ef-c0a80101-5-1(a)192.168.0.10 User: 7082@mydomain Contact: "" <sip:7082@192.168.0.10:5090;transport=udp;user=phone;fs_nat=yes;fs_path=sip%3A7082%4086.201.x.x%3A5090%3Btransport%3Dudp%3Buser%3Dphone> Agent: THOMSON ST2030 hw5 fw2.76 00-14-7F-00-4C-1A Status: Registered(UDP-NAT)(unknown) EXP(1907-10-11 18:28:34) EXPSECS(909113102) Ping-Status: Reachable Host: vpsxxxxxx.ovh.net IP: 86.201.x.x Port: 5090 Auth-User: 7082 Auth-Realm: mydomain MWI-Account: 7082@mydomain Any idea ? Best regards

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Configuring request parameters with mod_xml_curl
by Ben Hood 27 Jan '15

27 Jan '15

Hi, I'm using mod_xml_curl with FS 1.4.15 -1 and I was wondering if there is a way to configure the request parameters in FS to only send a subset of all available request parameters. My use case is that my web service responding to FS only uses two out of the many possible parameters that FS sends. Over time this is filling up the debug logs with lots of noise, so I was hoping to trim this down. Does anybody know of a way to do this? Thanks, Ben

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FreeSWITCH Week in Review (Master Branch) January 17th-23rd
by Ken Rice 26 Jan '15

26 Jan '15

New Post on freeswitch.org from kathleen check it out at http://ift.tt/1thCTTe FreeSWITCH Week in Review (Master Branch) January 17th-23rd Hello, again. This week in the FreeSWITCH master branch we had 21 commits. There were quite a few commits this week geared toward new features, including: adding switch_cache_db_create_schema() to test for SCF_AUTO_SCHEMAS flag, enable nat mode for verto when ext-rtp-ip is set, auto urlencode user portion of sip uri, uuid_media support for SRTP Call, and adding –enable-sytem-lua configure arg to allow building mod_lua against system lua and allow mod_lua to build against lua 5.1 or 5.2. These lua changes mean enabling the usage of system lua in preparation for removing the lua library from tree. New features that were added: 07d09b7 FS-7180 Add –enable-sytem-lua configure arg to allow building mod_lua against system lua and allow mod_lua to build against lua 5.1 or 5.2 [Jira: http://ift.tt/1z2ZB0V] 749ced5 FS-7180 More work toward add –enable-sytem-lua configure arg to allow building mod_lua against system lua and allow mod_lua to build against lua 5.1 or 5.2 [Jira: http://ift.tt/1z2ZB0V] 1d361b6 FS-7180 Let esl lua module build against lua 5.1 or 5.2 (requires newer swig) [Jira: http://ift.tt/1z2ZB0V] 861961b FS-7180 When using system lua, properly link against reanmed library versions on debian for mod_lua [Jira: http://ift.tt/1z2ZB0V] 8fc19a6 FS-7180 Properly build esl luamod when not using the –enable-system-lua configure arg [Jira: http://ift.tt/1z2ZB0V] 01dcb74 FS-7187 Add switch_cache_db_create_schema() to test for SCF_AUTO_SCHEMAS flag [Jira: http://ift.tt/1thCUXs] 1710214 Enable nat mode for verto when ext-rtp-ip is set 76370f4 Auto urlencode user portion of sip uri 83dd941 FS-7166 Uuid_media support for SRTP Calls [Jira: http://ift.tt/1z2ZDWC] 95a8efb Up the ice failover val to 3 sec Improvements in cross platform build supports: 90ab1d1 Fix cent5 build The following bugs were squashed: b744484 FS-7173 Work toward fixing a recording issues after an external call is transferred to an internal extension [Jira: http://ift.tt/1thCUXu] 15a7ff2 Fix hash dump gdb function f770b31 FS-7106 Fix a concurrency issue in caching of files from http urls [Jira: http://ift.tt/1BEamXd] 062ddcf FS-7174 Make sure not to leave any sessions readlocked in bridge_early_media=true in case one in the middle of the list is abandoned [Jira: http://ift.tt/1thCVdK] 90d3cb6 Fix media reload on verto and sip re-invites fc93895 FS-7173 More work toward fixing recording issues after an external call is transferred to an internal extension [Jira: http://ift.tt/1thCUXu] b37d071 FS-7186 Fix for enterprise bridging/originate issue [Jira: http://ift.tt/1z2ZDWF]

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Re: [Freeswitch-users] Fork( ) and Exec ( ) functions
by Abdul Hakeem 26 Jan '15

26 Jan '15

Hello again, The reason I ask is because I am trying to port FS to OSv. Unfortunately, Fork ( ) is unsupported in OSv. Regards, Abdul Hakeem From: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Moises Silva Sent: Tuesday, January 13, 2015 5:11 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Fork( ) and Exec ( ) functions On Tue, Jan 13, 2015 at 8:11 AM, Abdul Hakeem <alhakeem(a)gmail.com> wrote: Hello, I understand FS makes system calls for sending mails, voicemail and fax. Can anyone guide me on how to mitigate the load of fork ( ) and exec( ) on system calls & also, a list of functions which require FS to make system calls ? Not making much sense here. Everything in FS relies heavily on system calls as it's a multi-threaded-I/O-driven system for the most part. I think you're asking the wrong question.

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mod_snmp not loading
by Ahmed Habiba 26 Jan '15

26 Jan '15

Dears, kindly I tried to use mod_snmp I compile it using make && make install, however when I tried to load it I got the below message:, your kind usual support will be appreciated: 2015-01-24 15:05:58.673046 [CRIT] switch_loadable_module.c:1447 Error Loading module /usr/local/freeswitch/mod/mod_snmp.so **/usr/lib/libnetsnmpagent.so.15: undefined symbol: netsnmp_register_null_context** Thanks, Ahmed Habiba.

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