- FreeSWITCH-users - FreeSWITCH Mailing Lists

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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mod_event_multicast bind interface question
by Борисов, Дмитрий / Dmitriy Borisov 26 Jan '15

26 Jan '15

I have freeswitch installed on server with two etehrnet interfaces. My profiles are started on eth1, but multicast must be sent from eth0. Is it possible? And how? -- with best regards, Dmitriy Borisov

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callcenter mod rtmp cant receive calls...
by jdayola＠spingine.com 26 Jan '15

26 Jan '15

Hi! Im using mod callcenter. I have 3 agents on queue 600 and assigned them extensions 601,602 and 603. Everything was working fine when I tested it using xlite and zoiper. But on production we have our own softphone based on flash so we need to use rtmp. I already enabled rtmp but I don't receive calls on my extensions using rtmp. Like if I assign 601 and 602 to xlite and zoiper and 603 on our flash client only 601 and 602 can receive calls. Do I need to make a new dial plan if I use RTMP? Im still new to voip and Im using fusionpbx as my freeswitch gui. Regards, John D.

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Playing file using lua before bridge
by John Nash 25 Jan '15

25 Jan '15

I am trying to play some native files (Pre encoded) using lua script just after call is setup using Bridge (But before any answer). I need to use "Proxy_Media" (Want media to go through freeswitch but no codec translation). I tried using "bridge_pre_execute_aleg_app" but I do not hear sound files being played (Do not see any error message also). I have already tested lua script in some other test extension and it works fine. <extension name="main_route"> <condition field="destination_number" expression="^rb-(.+)$"> <action application="log" data="INFO progressing call..."/> <action application="set" data="proxy_media=true"/> <action application="set" data="bridge_pre_execute_aleg_app=lua"/> <action application="set" data="bridge_pre_execute_aleg_data=tt.lua ${credit_time}"/> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="execute_on_answer=sched_hangup +${credit_time}" /> <action application="bridge" data="sofia/$${domain}/${regex(${sip_req_uri}|^kb-(.+)$|%1)};fs_path=sip:${sip_received_ip}:${sip_received_port}"/> </condition> </extension>

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voicemail silence_detection is always detecting silence
by Raphael Lechner 25 Jan '15

25 Jan '15

Hi, I configured an extension that first call for some seconds a phone and if nobody is picking up, the caller is hearing a playback and can press 1 for leaving a voicemail and 2 to get redirected to a mobile phone. The Problem is that in my test environment with a SIP Provider everything works fine after tuning the voicemail.conf.xml to <param name="record-silence-threshold" value="200"/> <param name="record-silence-hits" value="10”/> But on the production machine we are using a patton as a ISDN Gateway and after pressing 1 for the voicemail I always got "Message is less than minimum record length: 3” on the calling phone. After that I can record a message and that works as expected. Is there a way do disable the silence_detection or any hint what I can change? I tried changing the silence-threshold to 1,50 and silence-hits to 300,30000 but nothing has changed Debug Log https://pastebin.freeswitch.org/23851 <https://pastebin.freeswitch.org/23851> The called python script: def handler(session, args): voicemail = args.split(' ')[0] dtmf_pressed = args.split(' ')[1] forward_number = args.split(' ')[2] callerid = session.getVariable("caller_id_number") callername = session.getVariable("caller_id_name").lstrip() if dtmf_pressed == '1': send_sms('377XXXXXXX’,’New Voicemail from %s %s' % (callername, callerid)) session.execute("export", "skip_greeting=true") session.execute("export", "skip_instructions=true") session.execute("answer") session.execute("voicemail", "default 192.168.17.252 10”) #session.execute("bridge", "loopback/app=voicemail:default %s %s" % (conf['network']['ip'],voicemail)) elif dtmf_pressed == '2': consoleLog( "info", "Call is forwarded to %s\n" % forward_number) session.transfer(forward_number, "XML", "default") else: consoleLog( "info", "DTMF received is %s and not 1 or 2.Hangup Call\n" % (dtmf_pressed)) session.hangup() Thank you, Raphael

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