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mod_vmd--can it do more than simply find VoiceMail Beeping?
by Matthew Fong 07 Apr '09

07 Apr '09

I'm doing some outbound dialing, and want to use mod_vmd to detect if a live person picks up or a voicemail picks up. I've read the wiki, and have been playing around with the dialplan implementation and the lua implementation, along with capturing the mod_vmdvmd::beep event. Using the examples on the wiki, I am able to call a number, sleep for 25 seconds, and mod_vmd usually detects a Beep (the answering machine beep right before you are to speak your message). My question is, is there a way to use mod_vmd to detect if an answering machine or human has picked up within the first 1-2 seconds after being answered? If so, can I get an example of how to set this up? my dialplan to test my lua implementation looks like  <extension name="vmdtest"> <condition field="destination_number" expression="^1986$"> <action application="answer"/> <action application="lua" data="matt_vmd.lua"/> <action application="hangup"/> </condition> </extension> and matt_vmd.lua looks like print ("--matt_vmd.lua START--") local human_detected = false; local voicemail_detected = false; function onInput(session, type, obj) if type == "dtmf" and obj['digit'] == '1' and human_detected == false then print('MATT--I detected a HUMAN'); human_detected = true; return "break"; end if type == "event" and voicemail_detected == false then print('MATT--I detected a VOICEMAIL'); voicemail_detected = true; return "break"; end end session:setInputCallback("onInput"); session:execute("vmd"); session:sleep(25000); print ("--matt_vmd.lua FINISHED--") Thanks. --matt

5 4

problems with Faktortel (AU) and multiple DID's and extensions
by David Robinson 07 Apr '09

07 Apr '09

I have been trying to setup 2 DID's to route to 2 extensions but whenever I try it, the second configured DID always routes to the first extension. In my public.xml I have the following: <include> <context name="public"> <extension name="DID 1"> <condition field="destination_number" expression="^(0746029000)$"> <action application="transfer" data="1000 XML default"/> </condition> </extension> <extension name="DID 2"> <condition field="destination_number" expression="^(0746029002)$"> <action application="transfer" data="1001 XML default"/> </condition> </extension> .... rest of file continues here ... While in my default.xml I have this: <include> <context name="default"> <extension name="David"> <condition field="destination_number" expression="^(0746029000)$" continue="on-true"> <action application="set" data="dialed_ext=$1" /> </condition> <condition field="destination_number" expression="^${caller_id_number}$"> <action application="set" data="voicemail_authorized=${sip_authorized}"/> <action application="answer"/> <action application="sleep" data="1000"/> <action application="voicemail" data="check default $${domain} ${dialed_ext}"/> <anti-action application="ring_ready"/> <anti-action application="set" data="call_timeout=10"/> <anti-action application="set" data="hangup_after_bridge=true"/> <anti-action application="set" data="continue_on_fail=true"/> <anti-action application="bridge" data="USER/1000@$${domain}"/> <anti-action application="answer"/> <anti-action application="sleep" data="1000"/> <anti-action application="voicemail" data="default $${domain} ${dialed_ext}"/> </condition> </extension> <extension name="Jake"> <condition field="destination_number" expression="^(0746029001)$" continue="on-true"> <action application="set" data="dialed_ext=$2" /> </condition> <condition field="destination_number" expression="^${caller_id_number}$"> <action application="set" data="voicemail_authorized=${sip_authorized}"/> <action application="answer"/> <action application="sleep" data="1000"/> <action application="voicemail" data="check default $${domain} ${dialed_ext}"/> <anti-action application="ring_ready"/> <anti-action application="set" data="call_timeout=10"/> <anti-action application="set" data="hangup_after_bridge=true"/> <anti-action application="set" data="continue_on_fail=true"/> <anti-action application="bridge" data="USER/1001@$${domain}"/> <anti-action application="answer"/> <anti-action application="sleep" data="1000"/> <anti-action application="voicemail" data="default $${domain} ${dialed_ext}"/> </condition> </extension> <extension name="outgoing - voicepulse"> <condition field="destination_number" expression="^(1{0,1}\d{10})$"> <action application="set" data="effective_caller_id_number=0746029000"/> <action application="bridge" data="sofia/gateway/voicepulse/$1"/> </condition> </extension> .. file continues here ... I got my new friend swk to try and diagnose the problem and using ngrep he found with ngrep that the incoming call to the second extension looked like this: U 203.161.130.133:5060 -> 10.0.0.12:5080 INVITE sip:gw+voicepulse@10.0.0.12:5080;transport=udp SIP/2.0..Via: SIP/2.0/UDP 203.161.130.133:5060;branch=z9hG4bK75f53071;rport..From: "0451282630" <sip: 0451282630(a)203.161.130.133>;tag=as555c5b50..To: <sip:gw+voicepulse@124.254.94.41:5080;transport=udp>..Contact: <sip:0451282630@203.161.130.133>..Call-ID: 7 47befb63a2def723e6796294853cc22@203.161.130.133..CSeq: 102 INVITE..User-Agent: Asterisk PBX..Max-Forwards: 70..Date: Tue, 07 Apr 2009 08:18:23 GMT..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY..Supported: replaces..Content-Type: application/sdp..Content-Length: 290....v=0..o=root 1244 12 44 IN IP4 203.161.130.133..s=session..c=IN IP4 203.161.130.133..t=0 0..m=audio 13806 RTP/AVP 18 3 101..a=rtpmap:18 G729/8000..a=fmtp:18 annexb=no..a=rtpmap :3 GSM/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-16..a=silenceSupp:off - - - -..a=ptime:20..a=sendrecv.. He says that the INVITE line should have a DNIS (not sure what that is) in that field to indicate which number to route it to but that for some reason, my provider (Faktortel in Australia) is not supplying that information. Does anyone know whether the problem is really at my provider's end or at my end, and if it's at my end, where ? thanks, pawz

1 0

Can anyone recommend any provisioning tools for use with FS?
by David Schwartz 07 Apr '09

07 Apr '09

Preferably GUI based. Thanks, David

1 0

Calls being cut off while recording a message
by Andy Ayers 07 Apr '09

07 Apr '09

Hi, I'm using freeswitch as a glorified answering machine. FS registers with a VOIP gateway and all calls into the gateway go through an ivr menu and are allowed to leave a message which gets recorded to a file. The FS box is behind a NAT firewall. Everything works fine except that intermittently, calls keep getting cut off after a number of seconds. I've attached a snapshot of the log at the point that the call gets cut off, can anyone suggest why this is happening or how I can prevent it? Many thanks Andy

2 12

Buzzing when people speak in conference
by Stromin Normin 07 Apr '09

07 Apr '09

Hi, I've been asked to do some testing on Freeswitch by work, we currently use Asterisk. I'm quite new to telephony so please go easy. I have FS setup on a windows box and at the moment I'm testing internal calls only, when I transfer calls or call extensions everything sounds great. The problem occurrs when I setup conferencing, people can log in ok and we can talk, the trouble is as people start to talk a buzzing sound is heard in the background, once the talking stops the buzzing stops. If the person goes on mute there is no buzzing. Hopefully this is enough info cheers for any help. _________________________________________________________________ Share your photos with Windows Live Photos – Free. http://clk.atdmt.com/UKM/go/134665338/direct/01/

10 14

Problem listening to PCMU and ul recordings
by Drew Ozier 07 Apr '09

07 Apr '09

Hi, I've been using the record_session feature and wish to use PCMU or ul audio format, but when I try to play back the audio in either format, it sounds high-pitch and fast as if it is playing back at 2x speed. I looked at the waveform recorded in PCMU and ul versus what it looks like when I record as wav, and it seems like it is only recording every-other sample (which would explain the pitch and speed). My vars.xml is set to PCMU as the global codec pref and the outbound codec pref. I am recording in stereo (one channel per leg of the call), but I'm not messing with any other recording parameters. Incidentally, the wav sounds just fine, but I'd prefer an 8-bit mulaw audio file, because I'm getting calls off a T1 (actually off of an AudioCodes Mediant 1000 that is converting the T1 to SIP for me), and I'd like to record precisely what is coming off the wire. I'd be happy to send any configuration files, I'm just currently at a loss for how to proceed. Thanks, Drew Ozier

1 0

libtool 2.2 patch
by Michael Jerris 06 Apr '09

06 Apr '09

I need some testers for systems using both libtool 2.2 and 1.5.x to confirm the following patch: http://jira.freeswitch.org/secure/attachment/11356/fs-r12922-libtool22.patch In order to test you will need to do a complete fresh checkout, apply this patch, then do a bootstrap, configure, etc. Please make sure both mod_spidermonkey and mod_shout both build AND load (you will need to edit modules.conf and modules.conf.xml for these modules) when you start freeswitch without any errors. Please post your test results to http://jira.freeswitch.org/browse/FSBUILD-82 Thanks Mike

3 3

MANDATORY_IE_MISSING and NORMAL_TEMPORARY_FAILURE using javascript in dialplan.
by mszlazak＠aol.com 06 Apr '09

06 Apr '09

On Windows XP/SP3 with FS trunk 12653M I get these errors using javascript in my dialplan: ? [MANDATORY_IE_MISSING] (see pastebin below) and/or with [CS_EXCHANGE_MEDIA] I get [NORMAL_TEMPORARY_FAILURE]? (not shown this time in pastebin) Here is the test javascript file: session.answer(); session.setVariable("choice", "demo"); If I remove session.answer() in the above test script then there is no problem but that doesn't always work with another scripts. Here is the related section of the dialplan. You dial into ext. 2222 and the problem happens at <action application="bridge" data="sofia/internal/${sip_to_user}@127.0.0.1:5068"/>. The variable "choice" is set to "demo" to get here. ? ??? <extension name="">??? ? ??? ??? <condition field="destination_number" expression="^2222$"> ? ??? ??? ???  ? ??? ??? ??? <action application="set" data="effective_caller_id_name=${caller_id_name}"/> ? ??? ??? ??? <action application="set" data="effective_caller_id_number=${caller_id_number}"/> ? ??? ??? ??? <action application="set" data="choice='not_saved'"/> ? ??? ??? ??? <action application="javascript" data="home_phone.js"/>? ??? ??? ??? ? ??? ??? ??? <action application="transfer" data="2223 XML default"/> ? ??? ??? </condition> ? ??? </extension> ? ??? ??? ??? <extension name="">??? ??? ??? ??? <condition field="destination_number" expression="^2223$"/> ? ??? ??? <condition field="${choice}" expression="^saved"> ? ??? ??? ??? <action application="set" data="api_hangup_hook=sched_api +60 none originate {id_name='${caller_id_name}',id_number=${caller_id_number},id_message=${uuid}}sofia/gateway/spa3102PSTN/12223334444@10.0.0.5:5061 GINO_ANS"/>??? ??? ??? ?? ??? ??? ??? <anti-action application="transfer" data="2224 XML default"/> ? ??? ??? </condition> ? ??? </extension> ? ??? ? ??? <extension name="">??? ? ??? ??? <condition field="destination_number" expression="^2224$"/> ?? ??? ??? <condition field="${choice}" expression="^demo"> ? ??? ??? ???  ? ??? ??? ??? <action application="set" data="effective_caller_id_name=${caller_id_name}"/> ? ??? ??? ??? <action application="set" data="effective_caller_id_number=${caller_id_number}"/> ?? ??? ??? ??? <action application="set" data="bypass_media=true"/> ??? ??? ??? <action application="set" data="hangup_after_bridge=true"/> ? ??? ??? ??? <action application="bridge" data="sofia/internal/${sip_to_user}@127.0.0.1:5068"/> ? ??? ??? ??? <anti-action application="hangup"/> ? ??? ??? </condition> ? ??? </extension> ? ??? <extension name=""> ? ??? ??? <condition field="destination_number" expression="^GINO_ANS$"> ? ??? ??? ??? <action application="answer"/> ? ??? ??? ??? <action application="javascript" data="notify_Gino.js '${id_name}' ${id_number} ${id_message}"/> ? ??? ??? ??? <action application="hangup"/> ? ??? ??? </condition> ? ??? </extension> I enabled SIP/Sofia tracing and "pastebinned" part of the output here: ?http://pastebin.freeswitch.org/8321 Thanks.

3 2

Native G729 file playback and recording in Windows
by Justin Miller 06 Apr '09

06 Apr '09

I know there is an implementation of this for linux. Does anyone have it working in Windows? I gave it a try, but had no luck. I can get individual G729 files to play through the dialplan, but I couldn't get voicemail to work. Justin

1 0

Re: [Freeswitch-users] Freeswitch not started OK at boottime?
by Fred 06 Apr '09

06 Apr '09

Brian West-3 > What run level are you starting freeswitch? 3 to 5, the default being 5 (it's the desktop version, hence starting with X): # cat /etc/inittab [...] id:5:initdefault: # chkconfig -l freeswitch freeswitch 0:off 1:off 2:off 3:on 4:on 5:on 6:off dujinfang > It seems started, I never used a suse, however, can you try this? #netstat -an | grep 8021 # netstat -an | grep 8021 tcp 0 0 127.0.0.1:8021 0.0.0.0:* LISTEN > Maybe FS started before network is ready. Check scripts in /etc/rc<your_run_level>.d/ or any equiv It looks ok: # ll /etc/rc.d/rc5.d/ [...] lrwxrwxrwx 1 root root 10 Jun 16 2008 S05network -> ../network [...] lrwxrwxrwx 1 root root 13 Mar 24 16:36 S12freeswitch -> ../freeswitch lrwxrwxrwx 1 root root 6 Jun 16 2008 S12xdm -> ../xdm lrwxrwxrwx 1 root root 8 Jun 16 2008 S14smbfs -> ../smbfs lrwxrwxrwx 1 root root 13 Jun 16 2008 S15cupsrenice -> ../cupsrenice If it's a Suse-specific issue, I'll go ask in a Suse forum and see if someone can figure it out. Thanks guys for the tips.

1 0