January 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Hangup on silence?
by Nicolas Brenner 04 Jan '10

04 Jan '10

Hi, is there a way to hangup after a certain amount of silence? my problem is that with regular PSTN calls I don't get a BYE from my provider until around 1 minute after the phone has been hanged up. This is pretty standard, and the provider is not at fault, but I would like to hangup the call after I detect a certain amount of silence time. The only thing I could find was wait_for_silence on the wiki, but that application only delays dialplan execution, and I would need something like: hangup_on_silence=30, where 30 means to hangup if 30 seconds of silence have passed. Thanks for your help! Nicolas

3 4

PHP ESL Problem
by Ron McLeod 04 Jan '10

04 Jan '10

Would someone please take a look at this simple PHP event socket script and tell me what I am doing wrong - or tell me that this could be a bug elsewhere? Any help would be appreciated. When I run the script without the call to execute(), everything seems fine. When I include the call to execute(), the calls to getType() return CUSTOM for a while, then later start to return the correct name. #!/usr/bin/php <?php require_once 'ESL.php'; $endPoint = 'sofia/internal/695%192.168.100.132'; $eventSocket = New ESLconnection('192.168.100.132', '8021', 'ClueCon'); $event = $eventSocket->events('plain', 'ALL'); // call endpoint, get uuid $event = $eventSocket->api('originate', $endPoint . ' &park'); $serializedEvent = explode("\n", $event->serialize()); foreach ($serializedEvent as $eventLine) { list($dummy, $uuid) = explode('+OK ', $eventLine); if ($uuid) { break; } } // play announcement to endpoint $event = $eventSocket->execute('playback', '/opt/ann/user-busy.wav', $uuid); // monitor events while (TRUE) { echo "getType: " . $event->getType() . "\n"; $serializedEvent = explode("\n", $event->serialize()); foreach ($serializedEvent as $eventLine) { list($header, $value) = explode(': ', $eventLine); if ($header == "Event-Name") { printf($eventLine . "\n"); } if ($header == "Content-Type") { printf($eventLine . "\n"); } } printf("\n"); $event = $eventSocket->recvEvent(); }?> Run without the call to execute(): ================================== getType: CUSTOM Content-Type: api/response getType: CHANNEL_CREATE Event-Name: CHANNEL_CREATE getType: CHANNEL_OUTGOING Event-Name: CHANNEL_OUTGOING getType: CHANNEL_ORIGINATE Event-Name: CHANNEL_ORIGINATE getType: CHANNEL_STATE Event-Name: CHANNEL_STATE getType: PRESENCE_IN Event-Name: PRESENCE_IN getType: CHANNEL_STATE Event-Name: CHANNEL_STATE getType: PRESENCE_IN Event-Name: PRESENCE_IN getType: CHANNEL_STATE Event-Name: CHANNEL_STATE getType: CALL_UPDATE Event-Name: CALL_UPDATE getType: CHANNEL_PROGRESS Event-Name: CHANNEL_PROGRESS getType: HEARTBEAT Event-Name: HEARTBEAT getType: HEARTBEAT Event-Name: RE_SCHEDULE getType: CALL_UPDATE Event-Name: CALL_UPDATE getType: CODEC Event-Name: CODEC getType: CODEC Event-Name: CODEC getType: CHANNEL_ANSWER Event-Name: CHANNEL_ANSWER getType: PRESENCE_IN Event-Name: PRESENCE_IN getType: API Event-Name: API getType: PRESENCE_IN Event-Name: PRESENCE_IN getType: CHANNEL_STATE Event-Name: CHANNEL_STATE getType: CHANNEL_EXECUTE Event-Name: CHANNEL_EXECUTE getType: CHANNEL_PARK Event-Name: CHANNEL_PARK getType: CHANNEL_HANGUP Event-Name: CHANNEL_HANGUP getType: CHANNEL_UNPARK Event-Name: CHANNEL_UNPARK getType: CHANNEL_EXECUTE_COMPLETE Event-Name: CHANNEL_EXECUTE_COMPLETE getType: PRESENCE_IN Event-Name: PRESENCE_IN getType: CHANNEL_STATE Event-Name: CHANNEL_STATE getType: PRESENCE_IN Event-Name: PRESENCE_IN getType: CHANNEL_HANGUP_COMPLETE Event-Name: CHANNEL_HANGUP_COMPLETE getType: CHANNEL_STATE Event-Name: CHANNEL_STATE getType: CHANNEL_DESTROY Event-Name: CHANNEL_DESTROY getType: CHANNEL_STATE Event-Name: CHANNEL_STATE Run with the call to execute(): =============================== getType: CUSTOM Content-Type: command/reply getType: CUSTOM Content-Type: text/event-plain Event-Name: CHANNEL_CREATE getType: CUSTOM Content-Type: text/event-plain Event-Name: CHANNEL_OUTGOING getType: CUSTOM Content-Type: text/event-plain Event-Name: CHANNEL_ORIGINATE getType: CUSTOM Content-Type: text/event-plain Event-Name: CHANNEL_STATE getType: CUSTOM Content-Type: text/event-plain Event-Name: PRESENCE_IN getType: CUSTOM Content-Type: text/event-plain Event-Name: CHANNEL_STATE getType: CUSTOM Content-Type: text/event-plain Event-Name: PRESENCE_IN getType: CUSTOM Content-Type: text/event-plain Event-Name: CHANNEL_STATE getType: CUSTOM Content-Type: text/event-plain Event-Name: CALL_UPDATE getType: CUSTOM Content-Type: text/event-plain Event-Name: CHANNEL_PROGRESS getType: CUSTOM Content-Type: text/event-plain Event-Name: CALL_UPDATE getType: CUSTOM Content-Type: text/event-plain Event-Name: CODEC getType: CUSTOM Content-Type: text/event-plain Event-Name: CODEC getType: CUSTOM Content-Type: text/event-plain Event-Name: CHANNEL_ANSWER getType: CUSTOM Content-Type: text/event-plain Event-Name: PRESENCE_IN getType: CUSTOM Content-Type: text/event-plain Event-Name: API getType: CUSTOM Content-Type: text/event-plain Event-Name: PRESENCE_IN getType: CUSTOM Content-Type: text/event-plain Event-Name: CHANNEL_STATE getType: CUSTOM Content-Type: text/event-plain Event-Name: CHANNEL_EXECUTE getType: CUSTOM Content-Type: text/event-plain Event-Name: CHANNEL_PARK getType: CHANNEL_EXECUTE Event-Name: CHANNEL_EXECUTE getType: CHANNEL_HANGUP Event-Name: CHANNEL_HANGUP getType: CHANNEL_EXECUTE_COMPLETE Event-Name: CHANNEL_EXECUTE_COMPLETE getType: COMMAND Event-Name: COMMAND getType: CHANNEL_UNPARK Event-Name: CHANNEL_UNPARK getType: CHANNEL_EXECUTE_COMPLETE Event-Name: CHANNEL_EXECUTE_COMPLETE getType: PRESENCE_IN Event-Name: PRESENCE_IN getType: CHANNEL_STATE Event-Name: CHANNEL_STATE getType: PRESENCE_IN Event-Name: PRESENCE_IN getType: CHANNEL_HANGUP_COMPLETE Event-Name: CHANNEL_HANGUP_COMPLETE getType: CHANNEL_STATE Event-Name: CHANNEL_STATE getType: CHANNEL_DESTROY Event-Name: CHANNEL_DESTROY getType: CHANNEL_STATE Event-Name: CHANNEL_STATE Thanks, Ron

3 2

system functions silently ignored
by Joseph L. Casale 04 Jan '10

04 Jan '10

What could cause this? I have a fax script which just stopped working. I see the logs display the complete diaplan which involves the system call, I can manually execute the very command from the cli, but during the call, it just skips the system application and hangups without printing anything in the logs? I even tried something simple like: <action application="system" data="touch /tmp/rxfax"/> Thanks, jlc

4 6

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.

2 2

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 -- --------------------------------------------------------------------------------------------------

3 2

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]

2 1

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

2 1

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

2 1

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? > > >

2 1

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