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Outbound SIP invites without codec list
by Liam Farr 25 May '11

25 May '11

Hi, My outbound SIP invites are going out without a codec list? Could someone point me in the right direction on how to set / insert this? send 972 bytes to udp/[1.2.3.4]:5060 at 00:28:52.144565: ------------------------------------------------------------------------ INVITE sip:0800000000@1.2.3.4:5060 SIP/2.0 Via: SIP/2.0/UDP 9.8.7.6;rport;branch=z9hG4bK2QeQtFFa7Z6Hr Max-Forwards: 69 From: "Liam - Test" <sip:12345678@9.8.7.6>;tag=ryFQea59e6Upe To: <sip:0800000000@1.2.3.4:5060> Call-ID: cec20463-0108-122f-689e-c99dbf5a2f43 CSeq: 12805602 INVITE Contact: <sip:12345678@9.8.7.6:5060> User-Agent: FreeSWITCH-mod_sofia/1.0.7-svn-exported Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 189 X-FS-Support: update_display Remote-Party-ID: "Liam - Test" <sip:12345678@9.8.7.6 >;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1306264686 1306264687 IN IP4 9.8.7.6 s=FreeSWITCH c=IN IP4 9.8.7.6 t=0 0 m=audio 18646 RTP/AVP 0 8 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 ------------------------------------------------------------------------ Thanks Liam

4 4

DTMF problems
by Boris Kovalenko 25 May '11

25 May '11

Hello! My network enviroment is: PRI --- Cisco 5350 --- FreeSwitch (Version 1.0.head (git-1c95ad9 2011-01-20 22-43-50 -0300)) --- AudioCodes MP-118FXS. There is an IVR on FreeSwitch where calling party may press a key for personal client line or wait for a secretary. Sometimes (as client said) the call to line 5 is placed to line 3 (as I undestand DTMF 5 is recognized as 3). How may I detect and solve the problem? Configurations below: Cisco 5350: dial-peer voice 4000 voip description OUTBOUND-CALLS-FROM-USI destination-pattern 50000#.T voice-class codec 1 session protocol sipv2 session target ipv4:82.193.138.187:5060 dtmf-relay rtp-nte fax-relay ecm disable fax rate 9600 fax protocol t38 ls-redundancy 5 hs-redundancy 0 fallback pass-through g711alaw no vad FreeSwitch: profile: <param name="dtmf-duration" value="2000"/>  (yes, it is commented) context: <extension name="ext_001"> <condition field="destination_number" expression="^(73435230100)$"> <action application="set" data="v_tagnet_ats_dstport=10005"/> <action application="set" data="v_ats_dstport=10005"/> <action application="set" data="import=v_tagnet_dstport,v_ats_dstpor t"/>  <action application="set" data="hangup_after_bridge=true"/>  <action application="set" data="ringback=$${ru-ring}"/> <action application="javascript" data="${scripts_dir}/common/set_dpe.js B ui.tagnet.cc 3 3"/> <action application="answer"/> <action application="sleep" data="1000"/> <action application="play_and_get_digits" data="1 1 1 5000 # /opt/fs/sounds/ntmjui/welcome.wav silence_stream://250 digits \d* 3000"/> <action application="execute_extension" data="ext_call_internal"/> <action application="bridge" data="${sofia_contact(101(a)ui.tagnet.cc)}" /> <action application="hangup"/> </condition> </extension> -- Regards, Boris

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Do ignore_display_updates=true only work on an Outgoing/B-Leg?
by Alan Frisch 25 May '11

25 May '11

Scenario is two FS boxes (FS1 and FS2) and OpenSIPs. FS1 sends Invite with "X-FS-Support: update_display" to FS2 via OpenSIPs FS2 configured with "global_setvar ignore_display_updates=yes" Even with ignore_display_updates enabled on FS2, FS2 responds with "X-FS-Display-Name" and "X-FS-Display-Numbers". FS1 is a box not under my administration, so is there a way to disable this response on FS2? I could easily strip the X-FS-Support header in OpenSIPs to prevent this, but seems like a kludgy way of getting things done. Is there something I am overlooking?

1 0

Setting gateway username dynamically
by Liam Farr 25 May '11

25 May '11

Hi, I have a sip provider who requires me to set the username to be the same as the outbound caller id for the gateway (trunk) to them. Is there a way to do this dynamically for each call? e.g. <include> <gateway name="SIP-Provider"> <param name="username" value="${outbound_caller_id_number}"/> <param name="password" value="password"/> <param name="realm" value="1.2.3.4"/> <param name="proxy" value="1.2.3.4"/> <param name="expire-seconds" value="800"/> <param name="register" value="false"/> <param name="retry-seconds" value="60"/> <param name="context" value="public"/> </gateway> </include> Cheers Liam

2 2

Scientific Linux 6 (RHEL6) and Tickless Kernel (and possibly Fedora releases)
by curriegrad2004 24 May '11

24 May '11

I was recently going through the configuration options on compiling a custom kernel for Scientific Linux 6 (aka RHEL 6 Clone) and I happened to find the kernel is being compiled to run tickless by default. Red Hat's documentation also confirms that the Kernel is indeed tickless by default: http://docs.redhat.com/docs/en-US/Red_Hat_Enterprise_Linux/6/html/Power_Man… Not knowing what a tickless meant, I went on and googled and found my answer to what a tickless kernel is. Then I decided to search up on tickless kernels with FreeSwitch and found out that 2 years ago somebody brought up the tickless kernel issue on this mailing list before and bkw did recommend in the past that users should disable the tickless feature of the kernel. Fast forwarding today is it still the case that when a user runs FreeSwitch, they should disable the tickless feature of the kernel? If so, I'd probably be adding the grub option of disabling the tickless feature with the kernel and adding this tidbit of information to the wiki.

1 0

Handling DTMF noinput and nomatch
by Yungwei Chen 24 May '11

24 May '11

Hi, The following javascript file asks callers to enter their 4 digit PINs. I am wondering if an event will be fired when a caller doesn't enter anything or enter just 3 digits. If not, what is the recommended way of handling those cases? Thanks. function on_dtmf(session, type, digits, arg) { dtmf_digits += digits.digit; return(false); } session.answer(); while (session.ready()) { dtmf_digits = ""; session.flushDigits(); session.speak("flite", "kal", 'please enter your 4 digit pin', on_dtmf); dtmf_digits = session.getDigits(4, "", 5000, 1000, 10000); console_log("pin=" + dtmf_digits + "\n"); }

2 1

Tuning Up Freeswitch
by Antonio Teixeira 24 May '11

24 May '11

Hello List. I'm currently integrating an IVR in python together with freeswitch using mod_python and ESL and my life has been well until ... The flow of calls went over 80 simultaneous calls. Now freeswitch starts sending packets with huge delays ( even when establishing the call , mainly the 200 ) and firing up the IVR with tons of delay up to 20 seconds. So i searched the wiki forums and mailing list: Put freeswitch on a diet , trimmed modules.conf Played with the ulimit stuff. Played with the IVRS to reduce load to a minimum and i was able to squeeze more 5 calls of performance. The problem is : Top shows top - 16:14:33 up 35 days, 8:15, 3 users, load average: 1.92, 1.76, 1.78 Tasks: 133 total, 1 running, 132 sleeping, 0 stopped, 0 zombie Cpu(s): 1.4%us, 3.3%sy, 0.0%ni, 94.6%id, 0.0%wa, 0.3%hi, 0.5%si, 0.0%st Mem: 8193336k total, 1639156k used, 6554180k free, 177208k buffers Swap: 19534904k total, 0k used, 19534904k free, 1062272k cached PID USER PR NI VIRT RES SHR S %CPU %MEM TIME+ COMMAND 31361 yadayada 20 0 716m 164m 9628 S 73 2.1 155:17.85 freeswitch Freeswitch goes up to 150 % and puff there goes the MOS down to 0. Some basic System Info : Debian 6.0 ( i heard the timming module is affected by Debian , but if the CPU % gets lower than 95% everything will be more stable) Python 2.5 2 x Intel(R) Xeon(R) CPU E5506 @ 2.13GHz 8 GB of Ram as you can see 94 % of the "Cpu Power" is sleeping :\ It appears freeswitch is only capable of using let's say "one cpu"/thread ?? Do you guys recommend simply starting more instances or redoing the IVR stuff. Hope you guys can help me out. Thanks António Teixeira

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Error playing TTS using flite
by Yungwei Chen 24 May '11

24 May '11

Hi, I am testing tts using flite against freeswitch (snapshot downloaded last week) on CentOS 5. I'm getting "Invalid speech module [cepstral]!" and "Invalid TTS module" errors when dialing 9911 from my SIP client. What am I missing here? Thanks. ... Dialplan: sofia/internal/1000(a)192.168.216.18 parsing [default->test1] continue=false Dialplan: sofia/internal/1000(a)192.168.216.18 Regex (PASS) [test1] destination_number(9911) =~ /^9911$/ break=on-false Dialplan: sofia/internal/1000(a)192.168.216.18 Action javascript(dtmf_test.js) --> Freeswitch figured out dtmf_test.js is the action to perform 2011-05-23 18:13:09.859007 [DEBUG] switch_core_state_machine.c:119 (sofia/internal/1000(a)192.168.216.18) State Change CS_ROUTING -> CS_EXECUTE 2011-05-23 18:13:09.859007 [DEBUG] switch_core_session.c:1114 Send signal sofia/internal/1000(a)192.168.216.18 [BREAK] 2011-05-23 18:13:09.859007 [DEBUG] switch_core_state_machine.c:364 (sofia/internal/1000(a)192.168.216.18) State ROUTING going to sleep 2011-05-23 18:13:09.859007 [DEBUG] switch_core_state_machine.c:325 (sofia/internal/1000(a)192.168.216.18) Running State Change CS_EXECUTE 2011-05-23 18:13:09.859007 [DEBUG] switch_core_state_machine.c:371 (sofia/internal/1000(a)192.168.216.18) State EXECUTE 2011-05-23 18:13:09.859007 [DEBUG] mod_sofia.c:240 sofia/internal/1000(a)192.168.216.18 SOFIA EXECUTE 2011-05-23 18:13:09.859007 [DEBUG] switch_core_state_machine.c:157 sofia/internal/1000(a)192.168.216.18 Standard EXECUTE EXECUTE sofia/internal/1000(a)192.168.216.18 set(open=true) 2011-05-23 18:13:09.859007 [DEBUG] mod_dptools.c:1060 sofia/internal/1000(a)192.168.216.18 SET [open]=[true] EXECUTE sofia/internal/1000(a)192.168.216.18 hash(insert/192.168.216.18-spymap/1000/399178b8-8592-11e0-ae8d-e9a6940f4ae1) EXECUTE sofia/internal/1000(a)192.168.216.18 hash(insert/192.168.216.18-last_dial/1000/9911) EXECUTE sofia/internal/1000(a)192.168.216.18 hash(insert/192.168.216.18-last_dial/global/399178b8-8592-11e0-ae8d-e9a6940f4ae1) EXECUTE sofia/internal/1000(a)192.168.216.18 set(RFC2822_DATE=Mon, 23 May 2011 18:13:09 -0500) 2011-05-23 18:13:09.859007 [DEBUG] mod_dptools.c:1060 sofia/internal/1000(a)192.168.216.18 SET [RFC2822_DATE]=[Mon, 23 May 2011 18:13:09 -0500] EXECUTE sofia/internal/1000(a)192.168.216.18 javascript(dtmf_test.js) 2011-05-23 18:13:09.879013 [DEBUG] sofia_glue.c:3034 AUDIO RTP [sofia/internal/1000(a)192.168.216.18] 192.168.216.18 port 31886 -> 192.168.216.9 port 5062 codec: 8 ms: 20 2011-05-23 18:13:09.879013 [DEBUG] switch_rtp.c:1636 Starting timer [soft] 160 bytes per 20ms 2011-05-23 18:13:09.879013 [DEBUG] sofia_glue.c:3296 Set 2833 dtmf send payload to 101 2011-05-23 18:13:09.879013 [DEBUG] sofia_glue.c:3301 Set 2833 dtmf receive payload to 101 2011-05-23 18:13:09.879013 [DEBUG] mod_sofia.c:681 Local SDP sofia/internal/1000(a)192.168.216.18: v=0 o=FreeSWITCH 1306160503 1306160504 IN IP4 192.168.216.18 s=FreeSWITCH c=IN IP4 192.168.216.18 t=0 0 m=audio 31886 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 2011-05-23 18:13:09.879013 [DEBUG] switch_core_session.c:707 Send signal sofia/internal/1000(a)192.168.216.18 [BREAK] 2011-05-23 18:13:09.879013 [DEBUG] switch_channel.c:2859 (sofia/internal/1000(a)192.168.216.18) Callstate Change RINGING -> ACTIVE 2011-05-23 18:13:09.879013 [NOTICE] mod_spidermonkey.c:2068 Channel [sofia/internal/1000(a)192.168.216.18] has been answered 2011-05-23 18:13:09.879013 [DEBUG] dtmf_test.js:34 sayivrmenu: menu=[mainmenu] validdigits=[0123#] 2011-05-23 18:13:09.879013 [DEBUG] sofia.c:4770 Channel sofia/internal/1000(a)192.168.216.18 entering state [completed][200] 2011-05-23 18:13:09.879013 [DEBUG] switch_ivr_play_say.c:244 Handle speak-text:[Press 0 for the Operator.] (en:en) 2011-05-23 18:13:09.879013 [ERR] switch_core_speech.c:61 Invalid speech module [cepstral]! -->cepstral is never installed. Why did Freeswitch try to use cepstral? 2011-05-23 18:13:09.879013 [ERR] switch_ivr_play_say.c:2361 Invalid TTS module! 2011-05-23 18:13:09.879013 [DEBUG] sofia.c:4770 Channel sofia/internal/1000(a)192.168.216.18 entering state [ready][200] 2011-05-23 18:13:09.939014 [DEBUG] switch_rtp.c:3104 Correct ip/port confirmed. 2011-05-23 18:13:15.979391 [DEBUG] switch_ivr_play_say.c:244 Handle speak-text:[Press 0 for the Operator.] (en:en) 2011-05-23 18:13:15.979391 [ERR] switch_core_speech.c:61 Invalid speech module [cepstral]! 2011-05-23 18:13:15.979391 [ERR] switch_ivr_play_say.c:2361 Invalid TTS module! 2011-05-23 18:13:22.079772 [DEBUG] switch_ivr_play_say.c:244 Handle speak-text:[Press 0 for the Operator.] (en:en) 2011-05-23 18:13:22.079772 [ERR] switch_core_speech.c:61 Invalid speech module [cepstral]! 2011-05-23 18:13:22.079772 [ERR] switch_ivr_play_say.c:2361 Invalid TTS module! During installatio, I uncommented the following in modules.conf so mod_flite is installed. mod_cepstral remains commented. asr_tts/mod_unimrcp asr_tts/mod_flite asr_tts/mod_pocketsphinx After installation, I uncommented mod_flite in /usr/local/freeswitch/conf/autoload_configs/modules.conf.xml  <load module="mod_flite"/>     Here's my dial-plan <extension name="test1"> <condition field="destination_number" expression="^9911$"> <action application="set" data="tts_engine=flite"/> <action application="set" data="tts_voice=kal"/> <action application="javascript" data="dtmf_test.js" /> </condition> </extension> dtmf_test.js is copied from http://wiki.freeswitch.org/wiki/Examples_ivrmenu_js to /usr/local/freeswitch/scripts folder. The phrase macro is also copied from that link to /usr/local/freeswitch/conf/lang/en/demo folder.

2 1

Read SIP Response Headers from Dialplan or Lua
by Eric Beard 24 May '11

24 May '11

Hello, I'm wondering if it's possible to read SIP response headers from the dial plan, or from Lua. One of my gateways returns a generic 500 for all kinds of different error conditions, and expects me to read variables in the sip response. e.g. SIP/2.0 500 Service Unavailable Via:.. From:... Etc: X-Custom-Var:val The value in the custom variable might be "busy", "bad number", etc. If I add something to the INVITE, I know that I can read it with channel variables "sip_h_X-Name", but I don't get the same thing for the response. Thanks, ----------------------- Eric Z. Beard, CTO Loop LLC w (877) 850-2010 x9249 m (727) 776-2768 eric(a)loopfx.com<mailto:eric@loopfx.com>

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My Freeswitch status
by Dome Charoenyost 24 May '11

24 May '11

10 million calls :) freeswitch@internal> status UP 0 years, 98 days, 16 hours, 45 minutes, 30 seconds, 743 milliseconds, 805 microseconds 10131867 session(s) since startup 123 session(s) 0/300 5000 session(s) max min idle cpu 0.00/70.00 Thank FS Dome C.

2 1