December 2008 - FreeSWITCH-users - FreeSWITCH Mailing Lists

LUA execute response question
by jonathan augenstine 30 Dec '08

30 Dec '08

I am developing a Freeswitch/LUA script. From the script, I have a requirement to retrieve information via HTTP from an app server. I have utilized the HTTP application from the Freeswitch CLI. It works great. My question is how can I call the "http get http://www......." from within the LUA script and retrieve the HTTP response? Is this feasible? Or is there a way to make an HTTP request directly from LUA? Jonathan

2 4

Freeswitch optimization as a registrar
by rod 30 Dec '08

30 Dec '08

Hi all, I know that freeswitch has not been designed as a pure sip proxy/registrar, but I'm wondering how many subscribers could be handled by FS. I setup the following test environment: - Kamailio 1.4.2 as the registrar - all invite requests are flowing through FS, even for a call between 2 registered subscribers. Many reasons for this: the calls CDR are centralized in the same format, I can easily add a billing ID to a call, proceed to recording, set the caller as anonymous if requested... - FS is used also as a SBC There is still a lot of work to do, mainly on the call forwarding feature and this is why I'm wondering (simply out of curiosity) what could have been achieved using only FS (easier to setup when only one equipment is involved :) ). I'd like to register 40 000 subscribers (if each user registers every 60s, you have approx 670 registration per second, this setup is working on Kamailio). I did the following to increase FS performance regarding registration: - put the directory containing users in a RAMDISK - put the db directory in a RAMDISK with this I was able to reach 190 registration per second (50 without the ramdisk) but for one SIP account, not too useful :p (for your information I see a huge improvement when switching from 1.0.1 phoenix: 150cps to FS svn 105xx: 190) When trying with 25000 SIP accounts, I got no more than 30cps. Then I tried to use the odbc mysql for registration, using this I was able to achieve 50cps. The mysql DB is not in a RAMDISK. For all these tests, the presence support has been disabled. As the IO performance seems to be a bottleneck, I'd like to know if there is a way to store the registration in memory only without database persistency. This thread is there only to share tips, not to complain about FS poor performance as a SIP registrar when compared to Kamailio. If I compare FS to a commercial SBC I'm using in production, I have to say that FS is really a great piece of software (lacks only statistics module, snmp, and heartbeat redundancy for failover). regards, rod

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how to use celt codec
by e schmidbauer 30 Dec '08

30 Dec '08

hi, great job with the celt module. im eager to use it but im not sure how. are there any sip clients that use the celt codec? or is there some other way to use the celt codec to play audio in a conference? thanks.

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FreeSWITCH News: New FreeSWITCH Client Available!
by Michael Collins 30 Dec '08

30 Dec '08

Hello FreeSWITCHers! Just a heads up, there are lots of cool things happening with FreeSWITCH. Please check out the latest here: http://freeswitch.org/node/156 Stay tuned for more news from the FreeSWITCH camp. -MC (mercutioviz)

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voicemail - Can't find user
by can_man＠gmx.de 30 Dec '08

30 Dec '08

Hello, I am trying to get voicemail to run through xml curl, but I get the following error: 2008-12-30 18:41:54 [WARNING] mod_voicemail.c:2737 voicemail_leave_main() Can't find user [315(a)192.168.178.22] In order to setup user 315 I reply the following to the "directory" request of xml curl: <user id="315" mailbox="315"> <params> <param name="password" value="1234"/> <param name="vm-password" value="0000"/> </params> <variables> <variable name="accountcode" value="315"/> <variable name="user_context" value="default"/> <variable name="vm_extension" value="315"/> <variable name="max_calls" value="1"/> <variable name="fail_over" value="415"/> <variable name="cringback" value="us-ring"/> </variables> </user> And in order to send the call to voicemail I do: <?xml version="1.0" encoding="UTF-8" standalone="no"?> <document type="freeswitch/xml"> <section name="dialplan" description="RE Dial Plan For FreeSwitch"> <context name="public"> <extension name="test10000"> <condition field="destination_number" expression="^(10000)$"> <action application="voicemail" data="default $${domain} 315"/> </condition> </extension> </context> </section> </document> Do I maybe have to add the user also at another location? Also, I read the following on the wiki: "I figured out that you can respond to both of these requests as follows. Probably the second one is looking for something different, but so far I just ignore it and throw out the same stuff." at http://wiki.freeswitch.org/wiki/Mod_xml_curl#bindings.3D.22directory.22 And I do the same, I respond always with the directory response above. Is there a better practice? It would be great if someone could point out my error. Thank you, Phil my voicemail conf looks like this: <configuration name="voicemail.conf" description="Voicemail"> <settings> </settings> <profiles> <profile name="default"> <param name="file-extension" value="wav"/> <param name="terminator-key" value="#"/> <param name="max-login-attempts" value="3"/> <param name="digit-timeout" value="10000"/> <param name="min-record-len" value="3"/> <param name="max-record-len" value="300"/> <param name="tone-spec" value="%(1000, 0, 640)"/> <param name="callback-dialplan" value="XML"/> <param name="callback-context" value="default"/> <param name="play-new-messages-key" value="1"/> <param name="play-saved-messages-key" value="2"/> <param name="main-menu-key" value="0"/> <param name="config-menu-key" value="5"/> <param name="record-greeting-key" value="1"/> <param name="choose-greeting-key" value="2"/> <param name="change-pass-key" value="6"/> <param name="record-name-key" value="3"/> <param name="record-file-key" value="3"/> <param name="listen-file-key" value="1"/> <param name="save-file-key" value="2"/> <param name="delete-file-key" value="7"/> <param name="undelete-file-key" value="8"/> <param name="email-key" value="4"/> <param name="pause-key" value="0"/> <param name="restart-key" value="1"/> <param name="ff-key" value="6"/> <param name="rew-key" value="4"/> <param name="record-silence-threshold" value="200"/> <param name="record-silence-hits" value="2"/> <param name="web-template-file" value="web-vm.tpl"/>    <param name="operator-extension" value="operator XML default"/> <param name="operator-key" value="9"/> <param name="vmain-extension" value="vmain XML default"/> <param name="vmain-key" value="*"/>   <email> <param name="template-file" value="voicemail.tpl"/> <param name="notify-template-file" value="notify-voicemail.tpl"/>  <param name="date-fmt" value="%A, %B %d %Y, %I %M %p"/> <param name="email-from" value="${voicemail_account}@${voicemail_domain}"/> </email>      </profile> </profiles> </configuration> the debug output: 2008-12-30 18:41:54 [INFO] mod_sofia.c:1272 sofia_receive_message() Asked to send early media by sofia/external/anonymous(a)sipgate.de 2008-12-30 18:41:54 [DEBUG] sofia_glue.c:497 sofia_glue_ext_address_lookup() STUN Success [89.49.116.108]:[61125] 2008-12-30 18:41:54 [DEBUG] sofia_glue.c:1825 sofia_glue_activate_rtp() AUDIO RTP [sofia/external/anonymous(a)sipgate.de] 192.168.178.22 port 25060 -> 217.10.77.21 port 57708 codec: 8 ms: 20 2008-12-30 18:41:54 [DEBUG] switch_rtp.c:859 switch_rtp_create() Starting timer [soft] 160 bytes per 20000ms 2008-12-30 18:41:54 [INFO] mod_sofia.c:1313 sofia_receive_message() Ring SDP: v=0 o=FreeSWITCH 1230597789 1230597790 IN IP4 89.49.116.108 s=FreeSWITCH c=IN IP4 89.49.116.108 t=0 0 m=audio 61125 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv 2008-12-30 18:41:54 [NOTICE] mod_sofia.c:1316 sofia_receive_message() Ring-Ready sofia/external/anonymous(a)sipgate.de! 2008-12-30 18:41:54 [NOTICE] mod_sofia.c:1316 sofia_receive_message() Pre-Answer sofia/external/anonymous(a)sipgate.de! 2008-12-30 18:41:54 [DEBUG] switch_core_session.c:510 switch_core_session_perform_receive_message() Send signal sofia/external/anonymous(a)sipgate.de [BREAK] 2008-12-30 18:41:54 [DEBUG] sofia.c:2542 sofia_handle_sip_i_state() Channel sofia/external/anonymous(a)sipgate.de entering state [early] 2008-12-30 18:41:54 [WARNING] mod_voicemail.c:2737 voicemail_leave_main() Can't find user [315(a)192.168.178.22] 2008-12-30 18:41:54 [DEBUG] switch_ivr_play_say.c:117 switch_ivr_phrase_macro() No language specified - Using [en] 2008-12-30 18:41:54 [DEBUG] switch_ivr_play_say.c:269 switch_ivr_phrase_macro() Handle play-file:[voicemail/vm-goodbye.wav] (en:en) 2008-12-30 18:41:54 [DEBUG] switch_ivr_play_say.c:932 switch_ivr_play_file() Codec Activated L16@8000hz 1 channels 20ms 2008-12-30 18:41:54 [DEBUG] switch_core_io.c:655 switch_core_session_write_frame() sofia/external/anonymous(a)sipgate.de receive message [SWITCH_MESSAGE_INDICATE_TRANSCODING_NECESSARY] 2008-12-30 18:41:55 [DEBUG] switch_ivr_play_say.c:1222 switch_ivr_play_file() done playing file 2008-12-30 18:41:55 [NOTICE] switch_core_state_machine.c:168 switch_core_standard_on_execute() Hangup sofia/external/anonymous(a)sipgate.de [CS_EXECUTE] [NORMAL_CLEARING] 2008-12-30 18:41:55 [DEBUG] switch_channel.c:1494 switch_channel_perform_hangup() Send signal sofia/external/anonymous(a)sipgate.de [KILL] 2008-12-30 18:41:55 [DEBUG] switch_core_session.c:806 switch_core_session_signal_state_change() Send signal sofia/external/anonymous(a)sipgate.de [BREAK] 2008-12-30 18:41:55 [DEBUG] switch_core_state_machine.c:442 switch_core_session_run() (sofia/external/anonymous(a)sipgate.de) State EXECUTE going to sleep 2008-12-30 18:41:55 [DEBUG] switch_core_state_machine.c:369 switch_core_session_run() (sofia/external/anonymous(a)sipgate.de) Running State Change CS_HANGUP 2008-12-30 18:41:55 [DEBUG] switch_core_state_machine.c:400 switch_core_session_run() (sofia/external/anonymous(a)sipgate.de) State HANGUP 2008-12-30 18:41:55 [DEBUG] mod_sofia.c:287 sofia_on_hangup() Channel sofia/external/anonymous(a)sipgate.de hanging up, cause: NORMAL_CLEARING 2008-12-30 18:41:55 [DEBUG] mod_sofia.c:361 sofia_on_hangup() Responding to INVITE with: 480 2008-12-30 18:41:57 [DEBUG] switch_core_state_machine.c:46 switch_core_standard_on_hangup() sofia/external/anonymous(a)sipgate.de Standard HANGUP, cause: NORMAL_CLEARING 2008-12-30 18:41:57 [DEBUG] switch_core_state_machine.c:400 switch_core_session_run() (sofia/external/anonymous(a)sipgate.de) State HANGUP going to sleep 2008-12-30 18:41:57 [DEBUG] switch_core_session.c:938 switch_core_session_thread() Session 2 (sofia/external/anonymous(a)sipgate.de) Locked, Waiting on external entities 2008-12-30 18:41:57 [NOTICE] switch_core_session.c:956 switch_core_session_thread() Session 2 (sofia/external/anonymous(a)sipgate.de) Ended 2008-12-30 18:41:57 [NOTICE] switch_core_session.c:958 switch_core_session_thread() Close Channel sofia/external/anonymous(a)sipgate.de [CS_HANGUP] -- Sensationsangebot verlängert: GMX FreeDSL - Telefonanschluss + DSL für nur 16,37 Euro/mtl.!* http://dsl.gmx.de/?ac=OM.AD.PD003K1308T4569a

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session_record post-processing
by Frank ＠ Impact 30 Dec '08

30 Dec '08

Maybe I am going about this all wrong. All I am trying to do is process a recording file of a session after either one of the legs hangs up and the call is over. I am just trying to convert the wav to mp3 and email it off. So I have a bash script to do this. The dialplan is simple enought <action application="set" data="RECORD_ANSWER_REQ=true"/> <action application="set" data="RECORD_STEREO=true"/> <action application="record_session" data="$${base_dir}/recordings/3329000_1.wav"/> <action application="bridge" data="${enum_auto_route}"/> <action application="hangup"/> using FS svn 10960 But nothing I have tried seems to get it done. I have tried to use api_hangup_hook to call a lua script. But the wav file is not yet closed for some reason yet and I cannot seem to close it in the lua script. Basically I get the same problem if I use 'system' with the api_hangup_hook to call the bash script to process the recording. I have tried to use the transfer application with the api_hangup_hook to allow me to stop_record_session and then a system call from another extension, but the transfer never happens on hangup. Am I missing a simple way to do this? Is there something similar to the 'h' extension in asterisk maybe?

4 5

Register Interval
by peder＠networkoblivion.com 30 Dec '08

30 Dec '08

What do most people use as a register interval for phones? On *, we always used 5 minutes and then had qualify setup, so we could keep track of the phones on a per minute basis as it "pinged" them every minute. FS doesn't do this to phones unless they are NAT'd, so if the reg is 5, we don't get any update for 5 minutes. I like the idea of checking my phones every minute so that I know if there is a problem right away when someone calls with an issue, rather than having to wait up to 5 minutes to see if it is still alive. Are most people using a small interval like 60 seconds? Or do they set it longer and just assume the phones are still alive in between registrations? Peder

1 0

Can't load module mod_xml_cdr.so
by fidibus83 30 Dec '08

30 Dec '08

Hello, I want to getting mod xml cdr working, but when I start freeswitch I get this Error: [CRIT] switch_loadable_module.c:756 switch_loadable_module_load_file() Error Loading module /usr/local/freeswitch/mod/mod_xml_cdr.so **/usr/lib/libnss3.so: undefined symbol: PR_UnloadLibrary** Why cant FS load the module? Thanks! Best regards

3 4

event_socket and stop_dtmf
by Peter P GMX 30 Dec '08

30 Dec '08

When I send a stop_dtmf command via event-socket, I get a channel_execute and a channel_execute_complete message back. However FS still accepts DTMFs and sends them via event-socket. In addition the other party will hear the DTMF. So I expect the stop_dtmf command is not really executed by FS. Here is the message I send: SendMsg call-command: execute execute-app-name: stop_dtmf execute-app-arg: true event-lock:true I send this command while I deliver a number of announcements to the user At Startup I get the following on the console 2008-12-29 22:48:05 [NOTICE] switch_loadable_module.c:238 switch_loadable_module_process() Adding Application 'stop_dtmf' 2008-12-29 22:48:05 [NOTICE] switch_loadable_module.c:238 switch_loadable_module_process() Adding Application 'stop_dtmf_generate' When I push stop_dtmf I get the following 2008-12-29 22:50:10 [DEBUG] switch_ivr.c:391 switch_ivr_parse_event() sofia/internal/1005(a)my.domain Command Execute stop_dtmf(true) What am I doing wrong here? Here's a console output when I push DTMF on either side after stop_dtmf has been pushed and the 2 call legs are bridged. 2008-12-29 23:13:39 [DEBUG] switch_rtp.c:1226 do_2833() Send start packet for [1] ts=417476340 dur=160/160/2000 seq=63290 2008-12-29 23:13:39 [DEBUG] switch_rtp.c:1168 do_2833() Send middle packet for [1] ts=417476340 dur=320/320/2000 seq=63291 2008-12-29 23:13:39 [DEBUG] switch_rtp.c:1168 do_2833() Send middle packet for [1] ts=417476340 dur=480/480/2000 seq=63292 2008-12-29 23:13:39 [DEBUG] switch_rtp.c:1168 do_2833() Send middle packet for [1] ts=417476340 dur=640/640/2000 seq=63293 2008-12-29 23:13:39 [DEBUG] switch_rtp.c:1168 do_2833() Send middle packet for [1] ts=417476340 dur=800/800/2000 seq=63294 2008-12-29 23:13:39 [DEBUG] switch_rtp.c:1168 do_2833() Send middle packet for [1] ts=417476340 dur=960/960/2000 seq=63295 2008-12-29 23:13:39 [DEBUG] switch_rtp.c:1168 do_2833() Send middle packet for [1] ts=417476340 dur=1120/1120/2000 seq=63296 2008-12-29 23:13:39 [DEBUG] switch_rtp.c:1168 do_2833() Send middle packet for [1] ts=417476340 dur=1280/1280/2000 seq=63297 2008-12-29 23:13:39 [DEBUG] switch_rtp.c:1168 do_2833() Send middle packet for [1] ts=417476340 dur=1440/1440/2000 seq=63298 2008-12-29 23:13:39 [DEBUG] switch_rtp.c:1168 do_2833() Send middle packet for [1] ts=417476340 dur=1600/1600/2000 seq=63299 2008-12-29 23:13:39 [DEBUG] switch_rtp.c:1168 do_2833() Send middle packet for [1] ts=417476340 dur=1760/1760/2000 seq=63300 2008-12-29 23:13:39 [DEBUG] switch_rtp.c:1168 do_2833() Send middle packet for [1] ts=417476340 dur=1920/1920/2000 seq=63301 2008-12-29 23:13:39 [DEBUG] switch_rtp.c:1168 do_2833() Send end packet for [1] ts=417476340 dur=2080/2080/2000 seq=63302 2008-12-29 23:13:39 [DEBUG] switch_rtp.c:1168 do_2833() Send end packet for [1] ts=417476340 dur=2080/2080/2000 seq=63303 2008-12-29 23:13:39 [DEBUG] switch_rtp.c:1168 do_2833() Send end packet for [1] ts=417476340 dur=2080/2080/2000 seq=63304 2008-12-29 23:13:44 [DEBUG] switch_rtp.c:1226 do_2833() Send start packet for [1] ts=292343252 dur=160/160/2000 seq=53073 2008-12-29 23:13:44 [DEBUG] switch_rtp.c:1168 do_2833() Send middle packet for [1] ts=292343252 dur=320/320/2000 seq=53074 2008-12-29 23:13:44 [DEBUG] switch_rtp.c:1168 do_2833() Send middle packet for [1] ts=292343252 dur=480/480/2000 seq=53075 2008-12-29 23:13:44 [DEBUG] switch_rtp.c:1168 do_2833() Send middle packet for [1] ts=292343252 dur=640/640/2000 seq=53076 2008-12-29 23:13:44 [DEBUG] switch_rtp.c:1168 do_2833() Send middle packet for [1] ts=292343252 dur=800/800/2000 seq=53077 2008-12-29 23:13:44 [DEBUG] switch_rtp.c:1168 do_2833() Send middle packet for [1] ts=292343252 dur=960/960/2000 seq=53078 2008-12-29 23:13:44 [DEBUG] switch_rtp.c:1168 do_2833() Send middle packet for [1] ts=292343252 dur=1120/1120/2000 seq=53079 2008-12-29 23:13:44 [DEBUG] switch_rtp.c:1168 do_2833() Send middle packet for [1] ts=292343252 dur=1280/1280/2000 seq=53080 2008-12-29 23:13:44 [DEBUG] switch_rtp.c:1168 do_2833() Send middle packet for [1] ts=292343252 dur=1440/1440/2000 seq=53081 2008-12-29 23:13:44 [DEBUG] switch_rtp.c:1168 do_2833() Send middle packet for [1] ts=292343252 dur=1600/1600/2000 seq=53082 2008-12-29 23:13:44 [DEBUG] switch_rtp.c:1168 do_2833() Send middle packet for [1] ts=292343252 dur=1760/1760/2000 seq=53083 2008-12-29 23:13:44 [DEBUG] switch_rtp.c:1168 do_2833() Send middle packet for [1] ts=292343252 dur=1920/1920/2000 seq=53084 2008-12-29 23:13:44 [DEBUG] switch_rtp.c:1168 do_2833() Send end packet for [1] ts=292343252 dur=2080/2080/2000 seq=53085 2008-12-29 23:13:44 [DEBUG] switch_rtp.c:1168 do_2833() Send end packet for [1] ts=292343252 dur=2080/2080/2000 seq=53086 2008-12-29 23:13:44 [DEBUG] switch_rtp.c:1168 do_2833() Send end packet for [1] ts=292343252 dur=2080/2080/2000 seq=53087 I am also wondering why I receive multiple events (15) for each dtmf pressed. I expect an echo floating back and forth and triggering dtmf, hein? Best regards Peter

2 3

xml lib curl - transfer isn't working
by can_man＠gmx.de 29 Dec '08

29 Dec '08

Hello, I am trying to replace some static settings with dynamic ones which are provided by a webserver. I can bridge calls that way, however I just can't get the following transfer to work. The transfer works when in public.xml and looks like this: <extension name="test9"> <condition field="destination_number" expression="^(10001)$"> <action application="set" data="domain_name=$${domain}"/> <action application="transfer" data="9999 XML default"/> </condition> </extension> The xml received by FS from the webserver looks like this: <context name="public"> <extension name="test9"> <condition field="destination_number" expression="^(10001)$"> <action application="set" data="domain_name=$${domain}"/> <action application="transfer" data="9999 XML default"/> </condition> </extension> </context> I have also tried without context name and extension name, but I got the same result. This is the console log output: 2008-12-25 21:27:27 [NOTICE] switch_channel.c:565 switch_channel_set_name() New Channel sofia/external/anonymous(a)sipgate.de [7281a542-d2c2-11dd-80f0-3fe65955e25b] 2008-12-25 21:27:27 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing anonymous->10001 in context public 2008-12-25 21:27:29 [CONSOLE] mod_xml_curl.c:236 xml_url_fetch() XML response is in /tmp/72829470-d2c2-11dd-80f0-3fe65955e25b.tmp.xml 2008-12-25 21:27:29 [NOTICE] switch_core_state_machine.c:168 switch_core_standard_on_execute() Hangup sofia/external/anonymous(a)sipgate.de [CS_EXECUTE] [NORMAL_CLEARING] 2008-12-25 21:27:29 [NOTICE] switch_core_session.c:956 switch_core_session_thread() Session 3 (sofia/external/anonymous(a)sipgate.de) Ended 2008-12-25 21:27:29 [NOTICE] switch_core_session.c:958 switch_core_session_thread() Close Channel sofia/external/anonymous(a)sipgate.de [CS_HANGUP] The 9999 extension in default looks like this: <extension name="hold_music"> <condition field="destination_number" expression="^9999$"/> <condition field="${sip_has_crypto}" expression="^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$"> <action application="answer"/> <action application="execute_extension" data="is_secure XML features"/> <action application="playback" data="$${hold_music}"/> <anti-action application="answer"/> <anti-action application="playback" data="$${hold_music}"/> </condition> </extension> Thank you very much for your help. Phil -- Sensationsangebot verlängert: GMX FreeDSL - Telefonanschluss + DSL für nur 16,37 Euro/mtl.!* http://dsl.gmx.de/?ac=OM.AD.PD003K1308T4569a

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