December 2011 - FreeSWITCH-users - FreeSWITCH Mailing Lists

LUA transfer result:0
by Anestis Mavro 24 Dec '11

24 Dec '11

Hi, I have tried two different ways of transfer in LUA: session:transfer(number,"XML","default") And Session:execute("transfer","12345 XML default") And I can't make it working. I get "switch_cpp.cpp:797 transfer result: 0" The same transfer from within the dialplan works fine The call is not answered before the transfer. Does anybody have an idea? Thanks Anestis

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how many luas scripts are running?
by David Villasmil 24 Dec '11

24 Dec '11

Hello gang, Is there any command that shows how many luas are running? Thanks David

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Help with a Cisco 7970 phone.
by Matt Piermarini 24 Dec '11

24 Dec '11

Hello all, I have been trying forever to get this Cisco 7970 to work properly with our freeswitch (latest git). The phone will register fine, and it will make outbound calls perfectly. The problem is with inbound calls to this phone, where FS tries to contact it on the wrong UDP port. The entire system here is on a local lan, and NAT has been disabled via --nonat cmd line FS parameter. Here is what it looks like when registered: sofia status profile internal reg Call-ID: 00181844-c24f0002-25f0dff2-c470b5a7(a)192.168.2.59 User: 15(a)x.y.com Contact: "user" <sip:15@192.168.2.59:5060;transport=udp;fs_nat=yes;fs_path=sip%3A15%40192.168.2.59%3A49320%3Btransport%3Dudp> Agent: Cisco-CP7970G/8.5.2 Status: Registered(UDP-NAT)(unknown) EXP(2011-12-22 21:31:47) EXPSECS(724) Host: x.y.com IP: 192.168.2.59 Port: 49165 Auth-User: 15 Auth-Realm: x.y.com FS tries to contact the phone using the fs_path, which is getting set incorrectly. I'm not why FS thinks the fs_path should be different, as it appears the Cisco is sending proper Contact header in the sip requests/responses. Also, we have a bunch of ATA's on the same lan which all work fine (meaning the fs_path is correct). I tried chaning the NDLB-force-rport param in the internal context, but didn't seems to change anything. Here are the SIP traces which show what's happening. Any ideas of what's going on would be appreciated. Thanks, Matt 192.168.2.59 is the cisco phone. 192.168.2.1 is FS (listening on port 5070). recv 713 bytes from udp/[192.168.2.59]:49320 at 02:28:01.712338: ------------------------------------------------------------------------ REGISTER sip:x.y.com SIP/2.0 Via: SIP/2.0/UDP 192.168.2.59:5060;branch=z9hG4bKd9e43394 From: <sip:15@x.y.com>;tag=00181844c24f0002a0bb4968-7b2b7dd4 To: <sip:15@x.y.com> Call-ID: 00181844-c24f0002-eb21e8a8-77681214(a)192.168.2.59 Max-Forwards: 70 Date: Fri, 23 Dec 2011 02:28:00 GMT CSeq: 101 REGISTER User-Agent: Cisco-CP7970G/8.5.2 Contact: <sip:15@192.168.2.59:5060;transport=udp>;+sip.instance="<urn:uuid:00000000-0000-0000-0000-00181844c24f>";+u.sip!model.ccm.cisco.com="30006" Supported: (null),X-cisco-xsi-7.0.1 Content-Length: 0 Reason: SIP;cause=200;text="cisco-alarm:20 Name=SEP00181844C24F Load=SIP70.8-5-2S Last=phone-keypad" Expires: 3600 send 689 bytes to udp/[192.168.2.59]:5060 at 02:28:01.715057: ------------------------------------------------------------------------ SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.2.59:5060;branch=z9hG4bKd9e43394 From: <sip:15@x.y.com>;tag=00181844c24f0002a0bb4968-7b2b7dd4 To: <sip:15@x.y.com>;tag=Qv2N52egZ2teH Call-ID: 00181844-c24f0002-eb21e8a8-77681214(a)192.168.2.59 CSeq: 101 REGISTER User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-8059cdc 2011-12-22 14-03-32 -0600 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces WWW-Authenticate: Digest realm="x.y.com", nonce="fae142ab-8b00-4ac7-89d7-04955e8714f6", algorithm=MD5, qop="auth" Content-Length: 0 recv 958 bytes from udp/[192.168.2.59]:49320 at 02:28:01.725802: ------------------------------------------------------------------------ REGISTER sip:x.y.com SIP/2.0 Via: SIP/2.0/UDP 192.168.2.59:5060;branch=z9hG4bK18d5a468 From: <sip:15@x.y.com>;tag=00181844c24f0002a0bb4968-7b2b7dd4 To: <sip:15@x.y.com> Call-ID: 00181844-c24f0002-eb21e8a8-77681214(a)192.168.2.59 Max-Forwards: 70 Date: Fri, 23 Dec 2011 02:28:00 GMT CSeq: 102 REGISTER User-Agent: Cisco-CP7970G/8.5.2 Contact: <sip:15@192.168.2.59:5060;transport=udp>;+sip.instance="<urn:uuid:00000000-0000-0000-0000-00181844c24f>";+u.sip!model.ccm.cisco.com="30006" Authorization: Digest username="15",realm="x.y.com",uri="sip:x.y.com",response="xxxxxxxxxxxxxxxxxxxxxxxxxxxxxx",nonce="fae142ab-8b00-4ac7-89d7-04955e8714f6",cnonce="4f2a64d4",qop=auth,nc=00000001,algorithm=MD5 Supported: (null),X-cisco-xsi-7.0.1 Content-Length: 0 Reason: SIP;cause=200;text="cisco-alarm:20 Name=SEP00181844C24F Load=SIP70.8-5-2S Last=phone-keypad" Expires: 3600 send 649 bytes to udp/[192.168.2.59]:5060 at 02:28:01.737460: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.2.59:5060;branch=z9hG4bK18d5a468 From: <sip:15@x.y.com>;tag=00181844c24f0002a0bb4968-7b2b7dd4 To: <sip:15@x.y.com>;tag=SeN78rgQSm7Kr Call-ID: 00181844-c24f0002-eb21e8a8-77681214(a)192.168.2.59 CSeq: 102 REGISTER Contact: <sip:15@192.168.2.59:5060;transport=udp>;expires=3600 Date: Fri, 23 Dec 2011 02:28:01 GMT User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-8059cdc 2011-12-22 14-03-32 -0600 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Content-Length: 0 *--------------- OK, so far so good.. phone is now registered.. Now look how FS tries to contact it.. sending notify's here: Note the port it tries to use, 49320 in this case.* send 1034 bytes to udp/[192.168.2.59]:49320 at 02:28:01.837149: ------------------------------------------------------------------------ NOTIFY sip:15@192.168.2.59:5060;transport=udp;fs_nat=yes;fs_path=sip:15%40192.168.2.59:49320%3Btransport%3Dudp SIP/2.0 Via: SIP/2.0/UDP 192.168.2.1:5070;rport;branch=z9hG4bKv92r0r53rQKNa Route: <sip:15@192.168.2.59:49320>;transport=udp Max-Forwards: 70 From: <sip:15@192.168.2.1>;tag=U07rcFjyK6KSF To: <sip:15@192.168.2.1> Call-ID: 93e179d6-a7b0-122f-d396-5254003b348b CSeq: 21967576 NOTIFY Contact: <sip:mod_sofia@192.168.2.1:5070> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-8059cdc 2011-12-22 14-03-32 -0600 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Event: message-summary Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Subscription-State: terminated;reason=timeout Content-Type: application/simple-message-summary Content-Length: 69 Messages-Waiting: no Message-Account: sip:15@x.y.com *The phone never responds, as FS is supposed to contact it on port 5060, but FS is trying to use the same port that phone sent is SIP messages on. Here is a sample INVITE:* send 1229 bytes to udp/[192.168.2.59]:49320 at 02:38:15.217775: ------------------------------------------------------------------------ INVITE sip:15@192.168.2.59:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.2.1:5070;rport;branch=z9hG4bK0D8U748HDUc0r Route: <sip:15@192.168.2.59:49320>;transport=udp Max-Forwards: 69 From: "x x x" <sip:11@192.168.2.1>;tag=63X9ZrB5K886e To: <sip:15@192.168.2.59:5060;transport=udp> Call-ID: 00e33727-a7b2-122f-d396-5254003b348b CSeq: 21967883 INVITE Contact: <sip:mod_sofia@192.168.2.1:5070> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-8059cdc 2011-12-22 14-03-32 -0600 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 204 X-FS-Support: update_display,send_info Remote-Party-ID: "x x x" <sip:11@192.168.2.1>;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1324545412 1324545413 IN IP4 192.168.2.1 s=FreeSWITCH c=IN IP4 192.168.2.1 t=0 0 m=audio 62482 RTP/AVP 18 0 8 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20

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bind digit action not working
by Rodney 23 Dec '11

23 Dec '11

Michael, I appreciate all the help you have given me but I am still struggling on the BDA. I put it exactly how you did and still it doesn't show in console when i press # and I get no response. condition destination_number ^501$ 1 action answer 2 action set conference_user_list=| 4 action bind_digit_action xfer,#,exec:transfer,759 XML default 5 action digit_action_set_realm xfer 6 action playback C:/Program Files/FreeSWITCH/sounds/en/us/callie/digits/8000/1.wav 8 action conference 501-127.0.0.1@default ________________________________ From: "freeswitch-users-request(a)lists.freeswitch.org" <freeswitch-users-request(a)lists.freeswitch.org> To: freeswitch-users(a)lists.freeswitch.org Sent: Friday, December 23, 2011 4:27 PM Subject: FreeSWITCH-users Digest, Vol 66, Issue 159 ----- Forwarded Message ----- Send FreeSWITCH-users mailing list submissions to freeswitch-users(a)lists.freeswitch.org To subscribe or unsubscribe via the World Wide Web, visit http://lists.freeswitch.org/mailman/listinfo/freeswitch-users or, via email, send a message with subject or body 'help' to freeswitch-users-request(a)lists.freeswitch.org You can reach the person managing the list at freeswitch-users-owner(a)lists.freeswitch.org When replying, please edit your Subject line so it is more specific than "Re: Contents of FreeSWITCH-users digest..." Today's Topics: 1. Re: transfer back to ivr after voicemail function (Michael Collins) 2. transfer from static conf to dynamic destination out of order. (Rodney) 3. Re: transfer from static conf to dynamic destination out of order. (Michael Collins) Just have more actions after voicemail in the dialplan. Also, you may wish to modify the voicemail_goodbye macro if you don't want it to say "thank you, goodbye" when exiting VM. -MC On Thu, Dec 22, 2011 at 12:49 PM, Rodney <notlikeme75(a)yahoo.com> wrote: I am trying to transfer someone back to the IVR after they leave a message for someone or check their own mailbox. currently it does a hangup and they have to call back to do more. is there a method for this? or is it hard coded in the "press # to continue portion of voice mail? > > > > >check voice mail > > >condition destination_number ^757$ 1 > action answer 3 > action sleep 1000 4 > action play_and_get_digits 4 4 3 5000 # checkvoicemail.wav ivr/ivr-that_was_an_invalid_entry.wav vmb \d+ 12 > action phrase spell,${vmb} 15 > action voicemail check default ${domain_name} ${vmb} > > > > >leave message > > > > > condition destination_number ^758$ 1 > action answer 3 > action sleep 1000 4 > action play_and_get_digits 4 4 3 5000 # checkvoicemail.wav ivr/ivr-that_was_an_invalid_entry.wav vmb \d+ 12 > action phrase spell,${vmb} 15 > action voicemail default ${domain_name} ${vmb} > >_________________________________________________________________________ >Professional FreeSWITCH Consulting Services: >consulting(a)freeswitch.org >http://www.freeswitchsolutions.com > >FreeSWITCH-powered IP PBX: The CudaTel Communication Server >http://www.cudatel.com > >Official FreeSWITCH Sites >http://www.freeswitch.org >http://wiki.freeswitch.org >http://www.cluecon.com > >FreeSWITCH-users mailing list >FreeSWITCH-users(a)lists.freeswitch.org >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >http://www.freeswitch.org > > MC, I will open up a jira for this possible bug. any help on the bind digit action would be great. i failed last time i tried. can bind digit action be single digit? this is what i am hoping for but i still want that digit to be "pound" i am thinking it wont work though. my extension into the static conference looks like this. where do put the bind digit action? before i have tried: backforward,7,exec:execute_extension,502 XML default (so pressing dtmf 7 while in static conf 501 would transfer to 502) but didnt work. didnt even show up as binding the "backforward" realm in console. condition destination_number ^501$ 1 action set conference_user_list=| 4 action playback C:/Program Files/FreeSWITCH/sounds/en/us/callie/digits/8000/1.wav 8 action answer 9 action conference 501-127.0.0.1@default any idea why it just hangs up and never gives me the extension? extension works if i put it on the ivr. http://pastebin.freeswitch.org/18064 ~rodney This looks like a possible bug. I can replicate on my system. I was able to use bind_digit_action to work around the issue. Open a Jira on this and report what you've got. Let me know if you need help trying out bind_digit_action (BDA). -MC On Fri, Dec 23, 2011 at 10:31 AM, Rodney <notlikeme75(a)yahoo.com> wrote: any idea why it just hangs up and never gives me the extension? extension works if i put it on the ivr. > > >http://pastebin.freeswitch.org/18064 > > > >~rodney > >_________________________________________________________________________ >Professional FreeSWITCH Consulting Services: >consulting(a)freeswitch.org >http://www.freeswitchsolutions.com > >FreeSWITCH-powered IP PBX: The CudaTel Communication Server >http://www.cudatel.com > >Official FreeSWITCH Sites >http://www.freeswitch.org >http://wiki.freeswitch.org >http://www.cluecon.com > >FreeSWITCH-users mailing list >FreeSWITCH-users(a)lists.freeswitch.org >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >http://www.freeswitch.org > > _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org Edit your conference.conf.xml and remove the # action in the caller controls. Then add something like this to the dialplan prior to going into the conference: <action application="bind_digit_action" data="xfer,#,exec:transfer,759 XML default"/> <action application="digit_action_set_realm" data="xfer"/> Note that you can change the BDA realm to something else if you want to change the meaning of the # key. That's why BDA is so awesome - you can do ANYTHING with it! -MC On Fri, Dec 23, 2011 at 1:05 PM, Rodney <notlikeme75(a)yahoo.com> wrote: MC, I will open up a jira for this possible bug. any help on the bind digit action would be great. i failed last time i tried. can bind digit action be single digit? this is what i am hoping for but i still want that digit to be "pound" i am thinking it wont work though. my extension into the static conference looks like this. where do put the bind digit action? > >before i have tried: > >backforward,7,exec:execute_extension,502 XML default (so pressing dtmf 7 while in static conf 501 would transfer to 502) but didnt work. didnt even show up as binding the "backforward" realm in console. > >condition destination_number ^501$ 1 > action set conference_user_list=| 4 > action playback C:/Program Files/FreeSWITCH/sounds/en/us/callie/digits/8000/1.wav 8 > action answer 9 > action conference 501-127.0.0.1@default > > > > > >any idea why it just hangs up and never gives me the extension? extension works if i put it on the ivr. > > >http://pastebin.freeswitch.org/18064 > > > > >~rodney > >This looks like a possible bug. I can replicate on my system. I was able to use bind_digit_action to work around the issue. Open a Jira on this and report what you've got. Let me know if you need help trying out bind_digit_action (BDA). > > >-MC > > >On Fri, Dec 23, 2011 at 10:31 AM, Rodney <notlikeme75(a)yahoo.com> wrote: > >any idea why it just hangs up and never gives me the extension? extension works if i put it on the ivr. >> >> >>http://pastebin.freeswitch.org/18064 >> >> >> >>~rodney >> >>_________________________________________________________________________ >>Professional FreeSWITCH Consulting Services: >>consulting(a)freeswitch.org >>http://www.freeswitchsolutions.com >> >>FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>http://www.cudatel.com >> >>Official FreeSWITCH Sites >>http://www.freeswitch.org >>http://wiki.freeswitch.org >>http://www.cluecon.com >> >>FreeSWITCH-users mailing list >>FreeSWITCH-users(a)lists.freeswitch.org >>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>http://www.freeswitch.org >> >> > >_______________________________________________ >FreeSWITCH-users mailing list >FreeSWITCH-users(a)lists.freeswitch.org >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >http://www.freeswitch.org > > > >_________________________________________________________________________ >Professional FreeSWITCH Consulting Services: >consulting(a)freeswitch.org >http://www.freeswitchsolutions.com > >FreeSWITCH-powered IP PBX: The CudaTel Communication Server >http://www.cudatel.com > >Official FreeSWITCH Sites >http://www.freeswitch.org >http://wiki.freeswitch.org >http://www.cluecon.com > >FreeSWITCH-users mailing list >FreeSWITCH-users(a)lists.freeswitch.org >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >http://www.freeswitch.org > > _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org

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CDRLogger
by sunsus 23 Dec '11

23 Dec '11

Hello Do somebody know were I can finde the source code of this project: http://wiki.freeswitch.org/wiki/JavaCDRLogger I can't finde it on git. Is there a backup of the svn? Regards Patrick -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/CDRLogger-tp7122659p7122659.h… Sent from the freeswitch-users mailing list archive at Nabble.com.

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transfer from static conf to dynamic destination out of order.
by Rodney 23 Dec '11

23 Dec '11

MC, I will open up a jira for this possible bug. any help on the bind digit action would be great. i failed last time i tried. can bind digit action be single digit? this is what i am hoping for but i still want that digit to be "pound" i am thinking it wont work though. my extension into the static conference looks like this. where do put the bind digit action? before i have tried: backforward,7,exec:execute_extension,502 XML default (so pressing dtmf 7 while in static conf 501 would transfer to 502) but didnt work. didnt even show up as binding the "backforward" realm in console. condition destination_number ^501$ 1 action set conference_user_list=| 4 action playback C:/Program Files/FreeSWITCH/sounds/en/us/callie/digits/8000/1.wav 8 action answer 9 action conference 501-127.0.0.1@default any idea why it just hangs up and never gives me the extension? extension works if i put it on the ivr. http://pastebin.freeswitch.org/18064 ~rodney This looks like a possible bug. I can replicate on my system. I was able to use bind_digit_action to work around the issue. Open a Jira on this and report what you've got. Let me know if you need help trying out bind_digit_action (BDA). -MC On Fri, Dec 23, 2011 at 10:31 AM, Rodney <notlikeme75(a)yahoo.com> wrote: any idea why it just hangs up and never gives me the extension? extension works if i put it on the ivr. > > >http://pastebin.freeswitch.org/18064 > > > >~rodney > >_________________________________________________________________________ >Professional FreeSWITCH Consulting Services: >consulting(a)freeswitch.org >http://www.freeswitchsolutions.com > >FreeSWITCH-powered IP PBX: The CudaTel Communication Server >http://www.cudatel.com > >Official FreeSWITCH Sites >http://www.freeswitch.org >http://wiki.freeswitch.org >http://www.cluecon.com > >FreeSWITCH-users mailing list >FreeSWITCH-users(a)lists.freeswitch.org >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >http://www.freeswitch.org > > _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org

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transfer back to ivr after voicemail function
by Rodney 23 Dec '11

23 Dec '11

I am trying to transfer someone back to the IVR after they leave a message for someone or check their own mailbox. currently it does a hangup and they have to call back to do more. is there a method for this? or is it hard coded in the "press # to continue portion of voice mail? check voice mail condition destination_number ^757$ 1 action answer 3 action sleep 1000 4 action play_and_get_digits 4 4 3 5000 # checkvoicemail.wav ivr/ivr-that_was_an_invalid_entry.wav vmb \d+ 12 action phrase spell,${vmb} 15 action voicemail check default ${domain_name} ${vmb} leave message condition destination_number ^758$ 1 action answer 3 action sleep 1000 4 action play_and_get_digits 4 4 3 5000 # checkvoicemail.wav ivr/ivr-that_was_an_invalid_entry.wav vmb \d+ 12 action phrase spell,${vmb} 15 action voicemail default ${domain_name} ${vmb}

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transfer fail from conf to conf / desination out of order
by Rodney 23 Dec '11

23 Dec '11

any idea why it just hangs up and never gives me the extension? extension works if i put it on the ivr. http://pastebin.freeswitch.org/18064 ~rodney

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Rent freeswitch box for production
by David Wafula 23 Dec '11

23 Dec '11

Hi Team, Anyone recommend where i can rent box for hosting freeswitch to be used as a conference server. I dont want a virtual box please. Regards, -- David Wafula

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Can't quite get call screening to work
by Phil Quesinberry 23 Dec '11

23 Dec '11

I'm trying to use the call screening example in the wiki and can't get FS to play the caller's name back to the destination extension. example here: http://wiki.freeswitch.org/wiki/FreeSwitch_Dialplan_XML#Example_13:_Call_Scr eening The initial announcement asking for the caller's name works fine: EXECUTE sofia/internal/102@192.168.1.6:5060 phrase(voicemail_record_name) 2011-12-07 14:52:13.715494 [DEBUG] mod_dptools.c:2362 Execute voicemail_record_name() lang 2011-12-07 14:52:13.715494 [DEBUG] switch_ivr_play_say.c:67 No language specified - Using [en] 2011-12-07 14:52:13.715494 [DEBUG] switch_ivr_play_say.c:244 Handle play-file:[voicemail/vm-record_name1.wav] (en:en) 2011-12-07 14:52:13.715494 [DEBUG] switch_ivr_play_say.c:1302 Codec Activated L16@8000hz 1 channels 20ms 2011-12-07 14:52:18.695891 [DEBUG] switch_ivr_play_say.c:1672 done playing file /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-record_name1.wav EXECUTE sofia/internal/102@192.168.1.6:5060 playback(tone_stream://%(500, 0, 640)) 2011-12-07 14:52:18.799791 [DEBUG] switch_ivr_play_say.c:1302 Codec Activated L16@8000hz 1 channels 20ms 2011-12-07 14:52:19.299814 [DEBUG] switch_ivr_play_say.c:1672 done playing file tone_stream://%(500, 0, 640) EXECUTE sofia/internal/102@192.168.1.6:5060 set(playback_terminators=#*0123456789) 2011-12-07 14:52:19.299814 [DEBUG] mod_dptools.c:1263 sofia/internal/102@192.168.1.6:5060 SET [playback_terminators]=[#*0123456789] Then the caller's name is recorded, and I've verified that the recording is indeed saved in /tmp: EXECUTE sofia/internal/102@192.168.1.6:5060 record(/tmp/102-name.wav 7 200 2) 2011-12-07 14:52:19.299814 [DEBUG] switch_ivr_play_say.c:585 Raw Codec Activated 2011-12-07 14:52:19.299814 [DEBUG] switch_core_codec.c:116 sofia/internal/102@192.168.1.6:5060 Push codec L16:70 . EXECUTE sofia/internal/102@192.168.1.6:5060 set(group_confirm_key=1) 2011-12-07 14:52:21.623965 [DEBUG] mod_dptools.c:1263 sofia/internal/102@192.168.1.6:5060 SET [group_confirm_key]=[1] EXECUTE sofia/internal/102@192.168.1.6:5060 set(fail_on_single_reject=true) 2011-12-07 14:52:21.623965 [DEBUG] mod_dptools.c:1263 sofia/internal/102@192.168.1.6:5060 SET [fail_on_single_reject]=[true] EXECUTE sofia/internal/102@192.168.1.6:5060 set(group_confirm_file=phrase:screen_confirm:/tmp/102-name.wav) 2011-12-07 14:52:21.623965 [DEBUG] mod_dptools.c:1263 sofia/internal/102@192.168.1.6:5060 SET [group_confirm_file]=[phrase:screen_confirm:/tmp/102-name.wav] EXECUTE sofia/internal/102@192.168.1.6:5060 set(continue_on_fail=true) 2011-12-07 14:52:21.623965 [DEBUG] mod_dptools.c:1263 sofia/internal/102@192.168.1.6:5060 SET [continue_on_fail]=[true] EXECUTE sofia/internal/102@192.168.1.6:5060 bridge(user/102) . Then when attempting to play back the Output from the console showing the error is here: 2011-12-07 14:52:26.348406 [ERR] switch_ivr_play_say.c:142 Can't find macro screen_confirm. 2011-12-07 14:52:26.348406 [WARNING] switch_ivr_play_say.c:339 Macro [screen_confirm]: '/tmp/102-name.wav' did not match any patterns 2011-12-07 14:52:26.348406 [ERR] switch_ivr_originate.c:219 sofia/internal/sip:102@192.168.1.4:5060 Error Playing File! The call goes right to voicemail once the destination extension attempts to answer it. Where are these macros supposed to be stored? Somewhere under /usr/local/freeswitch/conf/lang/en? Do I need to create a macro for screen_confirm or is it just named incorrectly or in the wrong place? Thanks, Phil Quesinberry Q Systems Engineering, Inc. Electronic Controls and Embedded Systems Development (410) 969-8002 http://www.qsystemsengineering.com

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