April 2009 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Hi Load, but calls still perfect
by Nik Middleton 07 Apr '09

07 Apr '09

Hi Guys, I'm no Linux guru, but today I inadvertently had 1000+ call attempts going through FS, load according to TOP was 16.5. Calls were still absolutely perfect. Can I throw out the rule book on load ? CPU was ~45% on each core. (dual) Regards,

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Skype interaction commands on skypiax
by Giovanni Maruzzelli 07 Apr '09

07 Apr '09

Hi all, background: mod_skypiax is Skype compatible endpoint that allows incoming and outbound calls to/from the Skype network and SkypeOut service. It's seen by FS like other endpoints, so you can originate from sofia, bridge to skypiax, and connect the call to a landline number via SkypeOut service, for eg. skypiax endpoint use a Skype client to interact with the Skype network (see the wiki page for more details http://wiki.freeswitch.org/wiki/Skypiax) The news are: now you can send commands to the skype client related to a skyiax interface, both from the FS command line and programmatically (socket/API/esl/whatever) http://wiki.freeswitch.org/wiki/Skypiax#API_and_CLI_Commands This allow you to use directly the entire power of the Skype API ( https://developer.skype.com/Docs/ApiDoc ), for eg to send chat messages, interact with the buddy list, etc etc. Typing "console loglevel 9" at the FS command line allows you to see the Skype API answers from the Skype client instance. So, in short: you bring loglevel to 9 (so you can see the Skype API messages going back and forth), you use "sk" or "skypiax" to send Skype API commands to the Skype client instance. This way you can prototype extensions to the current mod_skypiax, that can then be implemented in C directly into the mod_skypiax source code. Please, let me know of extensions you would like to be integrated into the mod_skypiax code ;-). Sincerely, Giovanni Maruzzelli ========================================= www.celliax.org via Pierlombardo 9, 20135 Milano Italy gmaruzz at celliax dot org Cell : +39-347-2665618 Fax : +39-02-87390039

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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. Incedentally, 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 precicely 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

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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

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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

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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

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

07 Apr '09

Preferably GUI based. Thanks, David

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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

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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/

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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