- FreeSWITCH-users - FreeSWITCH Mailing Lists

Was call answered or not?
by Sergey Okhapkin 04 Jan '10

04 Jan '10

Which channel variable accessible from mod_cdr_csv can be used to reliable find out if the call was answered or not? "billsec" can't be used - it is equal to 0 if the call has been answered for less than 1 second.

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jitter buffer effecting input callback in lua
by Tom Carlson 04 Jan '10

04 Jan '10

I have a very simple lua script (shown below my message) This script plays a greeting, lets the caller record a message, detecting when caller is done by sensing a keypress. It then plays the message back to the caller. This works perfectly, except the audio quality of the recorded message is less than I had hoped. To try to fix this, I have added a line to activate the jitter buffer. This single line keeps the script from detecting the dtmf tones that end the recording, so the script just stays locked in record mode forever, until you hang up. The log shows no problems. How can I activate the jitter buffer, and still detect dtmf events? Thanks for your help. Tom -- --------------------------------------------------------------------- function key_press(session, input_type, data, args) if input_type == "dtmf" then freeswitch.consoleLog("info", "Key pressed: " .. data["digit"]) return "break" end end session:setVariable("jitterbuffer_msec", "200"); if session:ready() then session:answer(); while (session:ready() == true) do session:setAutoHangup(false); session:sleep(1000); session:streamFile("/usr/local/freeswitch/sounds/en/us/callie/voicemail/8000/vm-record_message.wav"); session:setInputCallback("key_press", ""); session:recordFile("/tmp/blah.wav", 5000, 10, 10); -- pressing key ends the recording session:streamFile("/tmp/blah.wav"); end end -- --------------------------------------------------------------------------------------------------

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Dropped calls
by Mark Campbell-Smith 04 Jan '10

04 Jan '10

Hi! I just got a couple of dropped calls. Unfortunately I lost the sip traces, but I do have the debug logs... in both cases FS shows a Duplicate SDP received. I'm not sure if this is a cause - do these show anything to anyone as to why the calls dropped? FS version is FreeSWITCH Version 1.0.trunk (15490) Drop 1: 2010-01-02 17:18:50.806686 [DEBUG] switch_core_io.c:234 sofia/internal/2001@myddns.dydns.org:442 receive message [TRANSCODING_NECESSARY] 2010-01-02 17:18:50.826355 [DEBUG] switch_rtp.c:1972 Correct ip/port confirmed. 2010-01-02 17:18:51.645569 [DEBUG] sofia.c:3646 Channel sofia/internal/2001@myddns.dydns.org:442 entering state [ready][200] 2010-01-02 17:18:51.666115 [DEBUG] switch_core_session.c:704 Send signal sofia/internal/sip:1000@192.168.1.121:5060 [BREAK] 2010-01-02 17:18:51.666115 [DEBUG] switch_core_session.c:704 Send signal sofia/internal/2001@myddns.dydns.org:442 [BREAK] 2010-01-02 17:18:51.685576 [DEBUG] switch_ivr_bridge.c:122 sofia/internal/2001@myddns.dydns.org:442 receive message [DISPLAY] 2010-01-02 17:18:51.685576 [DEBUG] switch_ivr_bridge.c:122 sofia/internal/sip:1000@192.168.1.121:5060 receive message [DISPLAY] 2010-01-02 17:19:40.169692 [DEBUG] switch_rtp.c:2344 RTP RECV DTMF 1:404 2010-01-02 17:19:40.189926 [DEBUG] switch_rtp.c:1641 Send start packet for [1] ts=24065640 dur=160/160/404 seq=15337 2010-01-02 17:19:40.205843 [DEBUG] switch_rtp.c:1577 Send middle packet for [1] ts=24065640 dur=320/320/404 seq=15338 2010-01-02 17:19:40.225669 [DEBUG] switch_rtp.c:1577 Send end packet for [1] ts=24065640 dur=480/480/404 seq=15339 2010-01-02 17:19:40.225669 [DEBUG] switch_rtp.c:1577 Send end packet for [1] ts=24065640 dur=480/480/404 seq=15340 2010-01-02 17:19:40.225669 [DEBUG] switch_rtp.c:1577 Send end packet for [1] ts=24065640 dur=480/480/404 seq=15341 2010-01-02 17:19:48.942440 [DEBUG] sofia.c:3646 Channel sofia/internal/2001@myddns.dydns.org:442 entering state [calling][0] 2010-01-02 17:19:52.093594 [DEBUG] sofia.c:3646 Channel sofia/internal/2001@myddns.dydns.org:442 entering state [ready][200] 2010-01-02 17:19:52.093594 [DEBUG] sofia.c:3654 Duplicate SDP v=0 o=- 238296 238296 IN IP4 192.168.1.3 s=- c=IN IP4 192.168.1.3 t=0 0 m=audio 19428 RTP/AVP 2 0 8 4 96 97 98 100 101 a=rtpmap:2 G726-32/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:100 NSE/8000 a=fmtp:100 192-193 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 2010-01-02 17:19:59.713933 [DEBUG] switch_rtp.c:2344 RTP RECV DTMF 1:404 2010-01-02 17:19:59.725822 [DEBUG] switch_rtp.c:1641 Send start packet for [1] ts=24225880 dur=160/160/404 seq=16315 2010-01-02 17:19:59.745728 [DEBUG] switch_rtp.c:1577 Send middle packet for [1] ts=24225880 dur=320/320/404 seq=16316 2010-01-02 17:19:59.765676 [DEBUG] switch_rtp.c:1577 Send end packet for [1] ts=24225880 dur=480/480/404 seq=16317 2010-01-02 17:19:59.765676 [DEBUG] switch_rtp.c:1577 Send end packet for [1] ts=24225880 dur=480/480/404 seq=16318 2010-01-02 17:19:59.765676 [DEBUG] switch_rtp.c:1577 Send end packet for [1] ts=24225880 dur=480/480/404 seq=16319 2010-01-02 17:20:49.969621 [DEBUG] sofia.c:3646 Channel sofia/internal/2001@myddns.dydns.org:442 entering state [calling][0] 2010-01-02 17:20:54.036198 [DEBUG] sofia.c:3646 Channel sofia/internal/2001@myddns.dydns.org:442 entering state [terminating][503] 2010-01-02 17:20:54.036198 [NOTICE] sofia.c:4247 Hangup sofia/internal/2001@myddns.dydns.org:442 [CS_EXECUTE] [NORMAL_TEMPORARY_FAILURE] 2010-01-02 17:20:54.040419 [DEBUG] switch_channel.c:1912 Send signal sofia/internal/2001@myddns.dydns.org:442 [KILL] 2010-01-02 17:20:54.040419 [DEBUG] switch_core_session.c:982 Send signal sofia/internal/2001@myddns.dydns.org:442 [BREAK] Drop 2: 2010-01-02 17:10:42.177817 [DEBUG] switch_core_session.c:704 Send signal sofia/internal/sip:1000@192.168.1.121:5060 [BREAK] 2010-01-02 17:10:42.177817 [DEBUG] sofia.c:411 sofia/internal/sip:1000@192.168.1.121:5060 receive message [DISPLAY] 2010-01-02 17:10:42.177817 [DEBUG] sofia.c:3646 Channel sofia/internal/2001@myddns.dydns.org:442 entering state [completed][200] 2010-01-02 17:10:42.185735 [DEBUG] switch_core_session.c:643 Send signal sofia/internal/2001@myddns.dydns.org:442 [BREAK] 2010-01-02 17:10:42.189574 [NOTICE] switch_ivr_originate.c:2836 Channel [sofia/internal/2001@myddns.dydns.org:442] has been answered 2010-01-02 17:10:42.193802 [DEBUG] switch_channel.c:182 sofia/internal/2001@myddns.dydns.org:442 receive message [AUDIO_SYNC] 2010-01-02 17:10:42.193802 [DEBUG] switch_ivr_originate.c:2881 Originate Resulted in Success: [sofia/internal/sip:1000@192.168.1.121:5060] 2010-01-02 17:10:42.197882 [DEBUG] switch_channel.c:182 sofia/internal/sip:1000@192.168.1.121:5060 receive message [AUDIO_SYNC] 2010-01-02 17:10:42.201876 [DEBUG] switch_channel.c:182 sofia/internal/2001@myddns.dydns.org:442 receive message [AUDIO_SYNC] 2010-01-02 17:10:42.217738 [DEBUG] switch_ivr_originate.c:2881 Originate Resulted in Success: [sofia/internal/sip:1000@192.168.1.121:5060] 2010-01-02 17:10:42.221921 [DEBUG] switch_channel.c:182 sofia/internal/sip:1000@192.168.1.121:5060 receive message [AUDIO_SYNC] 2010-01-02 17:10:42.221921 [DEBUG] switch_channel.c:182 sofia/internal/2001@myddns.dydns.org:442 receive message [AUDIO_SYNC] 2010-01-02 17:10:42.237732 [DEBUG] switch_ivr_bridge.c:1004 sofia/internal/sip:1000@192.168.1.121:5060 receive message [BRIDGE] 2010-01-02 17:10:42.237732 [DEBUG] switch_core_session.c:643 Send signal sofia/internal/sip:1000@192.168.1.121:5060 [BREAK] 2010-01-02 17:10:42.241599 [DEBUG] switch_ivr_bridge.c:1011 sofia/internal/2001@myddns.dydns.org:442 receive message [BRIDGE] 2010-01-02 17:10:42.241599 [DEBUG] switch_core_session.c:643 Send signal sofia/internal/2001@myddns.dydns.org:442 [BREAK] 2010-01-02 17:10:42.241599 [DEBUG] switch_ivr_bridge.c:1055 (sofia/internal/sip:1000@192.168.1.121:5060) State Change CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA 2010-01-02 17:10:42.246345 [DEBUG] switch_core_session.c:982 Send signal sofia/internal/sip:1000@192.168.1.121:5060 [BREAK] 2010-01-02 17:10:42.246345 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/sip:1000@192.168.1.121:5060) Running State Change CS_EXCHANGE_MEDIA 2010-01-02 17:10:42.249596 [DEBUG] switch_core_state_machine.c:351 (sofia/internal/sip:1000@192.168.1.121:5060) State EXCHANGE_MEDIA 2010-01-02 17:10:42.249596 [DEBUG] mod_sofia.c:464 SOFIA LOOPBACK 2010-01-02 17:10:42.265702 [DEBUG] switch_core_io.c:234 sofia/internal/2001@myddns.dydns.org:442 receive message [TRANSCODING_NECESSARY] 2010-01-02 17:10:42.285925 [DEBUG] switch_rtp.c:1972 Correct ip/port confirmed. 2010-01-02 17:10:44.402422 [DEBUG] sofia.c:3646 Channel sofia/internal/2001@myddns.dydns.org:442 entering state [ready][200] 2010-01-02 17:10:44.413899 [DEBUG] switch_core_session.c:704 Send signal sofia/internal/sip:1000@192.168.1.121:5060 [BREAK] 2010-01-02 17:10:44.419743 [DEBUG] switch_core_session.c:704 Send signal sofia/internal/2001@myddns.dydns.org:442 [BREAK] 2010-01-02 17:10:44.425721 [DEBUG] switch_ivr_bridge.c:122 sofia/internal/sip:1000@192.168.1.121:5060 receive message [DISPLAY] 2010-01-02 17:10:44.433714 [DEBUG] switch_ivr_bridge.c:122 sofia/internal/2001@myddns.dydns.org:442 receive message [DISPLAY] 2010-01-02 17:11:44.822897 [DEBUG] sofia.c:3646 Channel sofia/internal/2001@myddns.dydns.org:442 entering state [calling][0] 2010-01-02 17:11:45.353266 [DEBUG] sofia.c:3646 Channel sofia/internal/2001@myddns.dydns.org:442 entering state [ready][200] 2010-01-02 17:11:45.353266 [DEBUG] sofia.c:3654 Duplicate SDP v=0 o=- 188941 188941 IN IP4 192.168.1.3 s=- c=IN IP4 192.168.1.3 t=0 0 m=audio 19424 RTP/AVP 2 0 8 4 96 97 98 100 101 a=rtpmap:2 G726-32/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:100 NSE/8000 a=fmtp:100 192-193 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 2010-01-02 17:12:44.834908 [DEBUG] sofia.c:3646 Channel sofia/internal/2001@myddns.dydns.org:442 entering state [calling][0] 2010-01-02 17:12:45.125887 [DEBUG] sofia.c:3646 Channel sofia/internal/2001@myddns.dydns.org:442 entering state [terminating][503] 2010-01-02 17:12:45.125887 [NOTICE] sofia.c:4247 Hangup sofia/internal/2001@myddns.dydns.org:442 [CS_EXECUTE] [NORMAL_TEMPORARY_FAILURE] 2010-01-02 17:12:45.125887 [DEBUG] switch_channel.c:1912 Send signal sofia/internal/2001@myddns.dydns.org:442 [KILL] 2010-01-02 17:12:45.125887 [DEBUG] switch_core_session.c:982 Send signal sofia/internal/2001@myddns.dydns.org:442 [BREAK]

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Re: [Freeswitch-users] Voicemail Question (using multiple servers)
by VM Knott 04 Jan '10

04 Jan '10

Bill, The "domain name" solution will not work for me. My FreeSWITCH servers are spread across multiple domain names, so I would prefer a solution that encompasses a more generic model. I'm guessing that my situation is more uncommon than I originally thought. I can setup something that is more specific to my architecture, and create a service-component that manages the central database to identify the greeting messages on all servers in the cluster. A less desirable approach, but in consideration of all of the other inherent features of FreeSWITCH, a small price to pay. - VMK ---------- Forwarded message ---------- From: "Bill W." <freeswitch(a)aastral.net> To: freeswitch-users(a)lists.freeswitch.org Date: Sun, 03 Jan 2010 10:24:49 -0500 Subject: Re: [Freeswitch-users] Voicemail Question (using multiple servers) Hey VM, I'm interested in your issue because I will need to implement this feature in probably 6 months or so. But I'm not currently familiar with the VM database records or how they're used. The weird thing is I'm using a shared sofia database for registrations across a cluster and it works fine. I know the IP address of the switch the UA registered to gets stored in the registration database, but any switch can use that registration record. Looking at the sql in the voicemail module, it shows a column for 'domain'. Is this where the IP address is being stored? If so, maybe you can find a way to change that to a domain name as Tony suggested. Is the domain name/ip address being used in the filesystem path? In any case I'm willing to help you solve this because I need to solve this issue as well. If you don't want to clog up the list with all the troubleshooting, we can take this off-list and then post our results to the thread when it's all done. Thanks, Bill

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Zap dialplan characteristics
by Joseph L. Casale 04 Jan '10

04 Jan '10

It seems it was permissions problems which were causing the audio issues for me, I was attempting to run a manually compiled instance of freeswitch with the stock init scripts. Apparently, setting the udev rules to freeswitch/daemon as fs runs won't work. I got it running finally tonight as a user and group 'freeswitch' but not till after I tried the zaptel release from the wiki's reco which didn't work until the perms issue was discovered. I am sure I can go back to using Digiums Dahdi package for Centos. So last question. W/ Asterisk, I had to answer the dahdi line so the far end didn't activate the call forward on no answer. Would it be safe in assuming this needs to be replicated here as well. If so, do I understand this right if I do this: <include> <extension name="openzap_in"> <condition field="source" expression="mod_openzap"> <action application="set" data="domain_name=$${domain}"/> <action application="ring_ready"/> <action application="transfer" data="2000 XML default"/> </condition> </extension> </include> So that I can ring the call group for its preset time which surely exceeds that of the call-forward from the telco? Thanks! jlc

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Re: [Freeswitch-users] Voicemail Question (using multiple servers)
by VM Knott 03 Jan '10

03 Jan '10

Bill, The original plan was to share the db across multiple switches. However, if I have to implement functionality to track records that identify a specific IP address for every switch in the cluster, I was thinking that a temporary work-around for our application would be a standard greeting for all mailboxes. We do not want the default TTS sounding greeting that identifies the mailbox number, so I was going to assign a standard greeting for all new mailboxes, and then if the owner submits a custom greeting to their mailbox, it will override the default. We can implement logic to synchronize the db, not a huge deal. But this just feels like I’m over-complicating the solution. - VMK ---------- Forwarded message ---------- From: "Bill W." <freeswitch(a)aastral.net> To: freeswitch-users(a)lists.freeswitch.org Date: Fri, 01 Jan 2010 01:31:14 -0500 Subject: Re: [Freeswitch-users] Voicemail Question (using multiple servers) So is the problem that you're having to replicate the voicemail database across switches in the cluster or is the problem the content of the entries in voicemail database? Because in your original post you're speaking of trying to share the voicemail db over NFS. Thanks, Bill VM Knott wrote: > Thank you Bill for the input, but changing how I access the data > source does not solve my problem. > I was hoping to avoid the management of IP Addresses for every > voicemail box on the system. > > Is there a way for me to set a default greeting to all voicemail boxes > globally, without having to go to a repository (regardless of means of > access) for each mailbox? > > >

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Mixing zap and sip users in the directory
by Joseph L. Casale 02 Jan '10

02 Jan '10

While trying to uncover my issues with no sound on a zap channel, somewhere in the archive I came across the mention of mixing analog and digital extensions in the directory, is there anything special that needs to be done for this? Also, to answer an incoming call from an fxo channel, can I simply transfer it: <include> <extension name="Openzap_in"> <condition field="source" expression="mod_openzap"> <action application="set" data="domain_name=$${domain}"/> <action application="transfer" data="2000 XML default"/> </condition> </extension> </include> Or do I have to answer and do other things? Thanks! jlc

1 0

User's Mailbox Password
by Sharad 02 Jan '10

02 Jan '10

Hi When a user changes his mailbox password from his phone using advance options, the corresponding XML does not show the new password. Can someone tell me what is the use of vm-password parameter which is shown in the XML of that user. regards Sharad -- View this message in context: http://n2.nabble.com/User-s-Mailbox-Password-tp4241899p4241899.html Sent from the freeswitch-users mailing list archive at Nabble.com.

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More Zap issues
by Joseph L. Casale 02 Jan '10

02 Jan '10

Now that the incoming zap is routed (and not hitting enum to get where it needs to be) I thought I would just send it to the example ivr (as I am remote) so I could see what happens at the cli. I can see it enter the public context then find the extension which sends it to 5000 XML default. I see all the prompts being played at the cli but there is nothing heard at the remote end? Any ideas what's wrong? During initial setup when it was hitting enum, people on the local end heard the ring and where able to answer and communicate with their sip phones? Thanks! jlc

1 0

Zap dialplan
by Joseph L. Casale 02 Jan '10

02 Jan '10

All the examples show bridging calls to an fx(s|o) port, but none show how to handle an incoming call from the pstn on an fxo port. How do you distinguish this call in your dialplan and begin routing it? Does fs just place it in the public context, and if so iirc pstn calls don't have a "destination_number" to match on, but a caller id only? Thanks, jlc

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