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default SIP registration timeout
by Madovsky 17 Jun '10

17 Jun '10

Is there any xml param for SIP registration timeout ? Thanks F

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Re: [Freeswitch-users] Question about execute_on_answer
by Davide Colombo 17 Jun '10

17 Jun '10

Hi, first of all, thanks for your support! I followed Steven's advice, using a script with exec in answer confirm to play a file and to read dtmf. In this way i'm able to connect caller party with called party, hang-up caller party, move to voicemail caller party, but with multiple calls, phones that don't answer, they continue to ring. To avoid this behaviour i tried in a different way. Calls start with a multiple originate: originate user/1001,user/1002,user/1003 exten-to-ivr When one user answers, he is transfered to an ivr where he can choose the action. In this way caller party remains in ringing state and other phones stop to ring. In exten-to-ivr caller party's uuid is passed to let connection with called party. Davide ----- Messaggio originale ----- Da: "Steven Ayre" <steveayre(a)gmail.com> A: freeswitch-users(a)lists.freeswitch.org Inviato: Martedì, 15 giugno 2010 11:06:37 Oggetto: Re: [Freeswitch-users] Question about execute_on_answer Read the "exec in answer confirm" section of http://wiki.freeswitch.org/wiki/Freeswitch_IVR_Originate#Answer_confirmation I think you would be able to do what you're asking within a script that's executed this way, playing the file and reading DTMF then doing logic based on the pressed key from within the script. Any scripting language should work - so lua etc as well as javascript. -Steve On 15 June 2010 09:46, Davide Colombo < dcolombo(a)voismart.it > wrote: Hi, using group_confirm_file and group_confirm_key i have only one choice, connect called party with calling party (1 dtmf maps with group_confirm_key). What i'd like to do when called party answers, it's: press 1 to connect with calling party, press 2 to hangup calling party, press 3 move calling party to voicemail. Additionally, making parellel dial in bridge, called phones stop to ring only when one of them digits the group_confirm_key and not (my purpose) when one of them answers. These are the reasons to transfer called channel to an ivr context using "execute_on_answer" variable. Best Regards ----- Messaggio originale ----- Da: "Michael Collins" < msc(a)freeswitch.org > A: freeswitch-users(a)lists.freeswitch.org Inviato: Martedì, 15 giugno 2010 0:37:18 Oggetto: Re: [Freeswitch-users] Question about execute_on_answer Aha! You are trying to do some sort of answer verification, correct? You do know that FS already has this feature, correct? Check it out: http://wiki.freeswitch.org/wiki/Channel_Variables#Answer_confirmation_varia… Make sure that you aren't duplicating functionality before continuing. -MC On Mon, Jun 14, 2010 at 3:12 PM, Davide Colombo < dcolombo(a)voismart.it > wrote: Hi, when 1019 answers, i transfer its channel (called channel party) to an IVR context using "execute_on_answer". In this context the called party (1019 in this case) can listen to a message where he can choose to connect/hang-up the calling party using dtmf tones. In the meanwhile i need that calling party remains in a ringing state. Best regards ----- Messaggio originale ----- Da: "Michael Collins" < msc(a)freeswitch.org > A: freeswitch-users(a)lists.freeswitch.org Inviato: Lunedì, 14 giugno 2010 23:46:42 Oggetto: Re: [Freeswitch-users] Question about execute_on_answer I'm not sure I follow what you're trying to accomplish... You call from x1000 to x1019... and what exactly do you want to have happen when 1019 answers? -MC On Mon, Jun 14, 2010 at 8:22 AM, Davide Colombo < dcolombo(a)voismart.it > wrote: Hi all, i'm trying to use "execute_on_answer" channel variable with a transfer before a bridge: <action application="export" data="nolocal:execute_on_answer=transfer myivrpresentation XML default"/> <action application="bridge" data="user/1019@${domain_name}"/> When i call 1019 from 1000 and 1019 answers, i have this situation: 1019 is transfered to myivrpresentation but in console i can see this error 2010-06-14 16:57:51.053528 [DEBUG] switch_core_codec.c:146 sofia/internal/ 1000(a)fs2-devel.voismart.net Restore previous codec PCMU:0. 2010-06-14 16:57:51.053528 [DEBUG] switch_ivr_originate.c:3310 Originate Resulted in Error Cause: 487 [ORIGINATOR_CANCEL] 2010-06-14 16:57:51.053528 [ERR] switch_ivr_originate.c:2491 Cannot create outgoing channel of type [user] cause: [ORIGINATOR_CANCEL] 2010-06-14 16:57:51.053528 [DEBUG] switch_ivr_originate.c:3310 Originate Resulted in Error Cause: 487 [ORIGINATOR_CANCEL] while 1000 goes to next action after bridge in the same extension. Is it possible to transfer called channel (when called party answers) to a particular extension and leave caller channel in ringing state? Best Regards _______________________________________________ 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 _______________________________________________ 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 _______________________________________________ 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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Aastra SCA Woes, Any Updates?
by Wes Pearce 17 Jun '10

17 Jun '10

Hello Users, I've got a dozen or so Aastra 6757i phones hooked up to FreeSWITCH in the office here. I've been trying to get SCA working unsuccessfully. While I was digging around, I found this thread: http://freeswitch-users.2379917.n2.nabble.com/Aastra-and-SCA-td4995252.html Which describes my exact problem. SCA works neatly on outbound calls, and SCA works on incoming calls until the phone is answered... at which point the call dissapears from the other lines. Brian West mentioned this has something to do with Aastras inconsistent call-info headers. I've been speaking with a support representative from Aastra named Brian Epps, who says their engineers are unaware of any SCA problems. Here are the appropriate SIP traces: incoming call: http://pastebin.com/QK9K3KL9 outgoing call: http://pastebin.com/zzxbm9mq There are call info headers in both traces. It's just the call-info headers in the incoming trace look incorrect. I'm wondering if there has been any movement on this, or if I've just screwed something up somewhere. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Aastra-SCA-Woes-Any-Updates-t… Sent from the freeswitch-users mailing list archive at Nabble.com.

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Issue with an IVR-menu
by Milena 17 Jun '10

17 Jun '10

Hello, Does anybody know why would freeswitch say ... switch_ivr_menu.c:851 Unable to build xml menu ... mod_dptools.c:1264 Unable to create menu when trying to call an IVR that was working fine before I updated this morning? ... and how to fix it? ;) http://pastebin.freeswitch.org/13209 thank you

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How to enable mod_skypopen without sound card
by k xd 17 Jun '10

17 Jun '10

Hi, I setup freeswitch in a window server and then startup 2 skype clients, however when I call registered skype user, I got this message "Contact can only receive IMs". The server doesn't have sound card. So I guess that when registered skype users try to transfer the incoming call, because of no sound card it terminated the call directly. Does anyone know how to solve this issue? Thanks, Will

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Javascript session.sayphrase()
by David Swardstrom 17 Jun '10

17 Jun '10

I have a need to provide date & time to a caller. The current application allows for interruption via a DTMF key. It would be nice to use session.sayphrase() but the Wiki seems to indicate that this does not allow operation in an interruptable way. Is this correct or do I need to go to lower levels and perhaps change xml files for the "Phrases Section"? so that a play can (optionally) use the dtmf detection and callback arguments? Regards, Paul David Swardstrom

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T38 unsuccessful: result (48) Disconnected after permitted retries
by samuel 17 Jun '10

17 Jun '10

hi all, recently testing T.38 unsuccesfully with freeswitch head version (FreeSWITCH Version 1.0.head (git-) ) on a testing server (2.6.26-2-xen-amd64) remote SDP: v=0 o=root 6039 6040 IN IP4 A.B.C.D s=session c=IN IP4 A.B.C.D t=0 0 m=image 19423 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:9600 a=T38FaxFillBitRemoval:0 a=T38FaxTranscodingMMR:0 a=T38FaxTranscodingJBIG:0 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:72 a=T38FaxMaxDatagram:72 a=T38FaxUdpEC:t38UDPRedundancy freeswitch answer: v=0 o=FreeSWITCH 1276579579 1276579581 IN IP4 W.X.Y.Z s=FreeSWITCH c=IN IP4 W.X.Y.Z t=0 0 m=image 22230 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxFillBitRemoval a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:2000 a=T38FaxMaxDatagram:400 a=T38FaxUdpEC:t38UDPRedundancy a=T38VendorInfo:0 0 0 some debug: 2010-06-15 13:37:13.234961 [DEBUG] mod_spandsp_fax.c:271 FLOW T.30 T4 expired in phase T30_PHASE_B_RX, state 17 2010-06-15 13:37:13.234961 [DEBUG] mod_spandsp_fax.c:271 FLOW T.30 Too many retries. Giving up. 2010-06-15 13:37:13.234961 [DEBUG] mod_spandsp_fax.c:271 FLOW T.30 Changing from phase T30_PHASE_B_RX to T30_PHASE_D_TX 2010-06-15 13:37:13.234961 [DEBUG] mod_spandsp_fax.c:271 FLOW T.38T Set rx type 0 2010-06-15 13:37:13.234961 [DEBUG] mod_spandsp_fax.c:271 FLOW T.38T Set tx type 4 2010-06-15 13:37:13.234961 [DEBUG] mod_spandsp_fax.c:271 FLOW T.30 Changing from state 17 to 3 2010-06-15 13:37:13.234961 [DEBUG] mod_spandsp_fax.c:271 FLOW T.30 Tx: DCN with final frame tag 2010-06-15 13:37:13.234961 [DEBUG] mod_spandsp_fax.c:271 FLOW T.30 Tx: ff 13 fa 2010-06-15 13:37:14.406960 [DEBUG] mod_spandsp_fax.c:271 FLOW T.30 Send complete in phase T30_PHASE_D_TX, state 3 2010-06-15 13:37:14.406960 [DEBUG] mod_spandsp_fax.c:271 FLOW T.30 Send complete in phase T30_PHASE_D_TX, state 3 2010-06-15 13:37:14.406960 [DEBUG] mod_spandsp_fax.c:271 FLOW T.30 Disconnecting 2010-06-15 13:37:14.406960 [DEBUG] mod_spandsp_fax.c:271 FLOW T.30 Changing from phase T30_PHASE_D_TX to T30_PHASE_E 2010-06-15 13:37:14.406960 [DEBUG] mod_spandsp_fax.c:271 FLOW T.38T Set rx type 0 2010-06-15 13:37:14.406960 [DEBUG] mod_spandsp_fax.c:271 FLOW T.38T Set tx type 1 2010-06-15 13:37:14.406960 [DEBUG] mod_spandsp_fax.c:271 FLOW T.30 Changing from state 3 to 2 2010-06-15 13:37:15.634931 [DEBUG] mod_spandsp_fax.c:271 FLOW T.30 Send complete in phase T30_PHASE_E, state 2 2010-06-15 13:37:15.634931 [DEBUG] mod_spandsp_fax.c:301 ============================================================================== 2010-06-15 13:37:15.634931 [DEBUG] mod_spandsp_fax.c:313 Fax processing not successful - result (48) Disconnected after permitted retries. 2010-06-15 13:37:15.634931 [DEBUG] mod_spandsp_fax.c:318 Remote station id: 2010-06-15 13:37:15.634931 [DEBUG] mod_spandsp_fax.c:319 Local station id: SpanDSP Fax Ident 2010-06-15 13:37:15.634931 [DEBUG] mod_spandsp_fax.c:320 Pages transferred: 0 2010-06-15 13:37:15.634931 [DEBUG] mod_spandsp_fax.c:322 Total fax pages: 0 2010-06-15 13:37:15.634931 [DEBUG] mod_spandsp_fax.c:323 Image resolution: 0x0 2010-06-15 13:37:15.634931 [DEBUG] mod_spandsp_fax.c:324 Transfer Rate: 14400 2010-06-15 13:37:15.634931 [DEBUG] mod_spandsp_fax.c:326 ECM status off I've starting learning about T3X and freswitch so I would like whether someone can point me to any doc or any configuration parameter that can solve the issue or at least debug it. Thanks in advance, Samuel

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Special Announcement: Be On The Community Conf Call Tomorrow!
by Michael Collins 17 Jun '10

17 Jun '10

Hey everyone! Brian West has an announcement for everyone so be sure to be on the community conference call tomorrow! The details are in the usual spot on the wiki: http://wiki.freeswitch.org/wiki/FS_weekly_2010_06_16 We have updates on the FS project plus a followup presentation by DRK on using mod_managed for CDRs and billing, etc. Talk to you tomorrow! -Michael

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Issue in fifo
by Abubacker siddiq 17 Jun '10

17 Jun '10

Dear all, I am facing problem in fifo while running that using outbound sockets in an async mode, it seems like connecting with the agents but no voice trasmission happens between them , please let me know if some body faced this issue already , and also give me a suggestion whether it is ok to run the fifo using outbound sockets. my main requirement is to play a file with a regular intervals for a waiting customers , Thanks in Advance !

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How to continue call even if recording fails?
by Maciej Aniserowicz 17 Jun '10

17 Jun '10

Hi, I'm recording sessions by sending uuid_record command to FS after calls are bridged. It works fine, but problems start when i'm sending a wrong/nonexistent directory to create the recorded file in. FS shows error (which is expected and fine): [ERR] mod_sndfile.c:195 Error Opening File [c:/FreeSWITCH/reco/rec_c1e3df00-dda2-4320-8900-4853a216f19b.wav] [System error : The system cannot find the path specified.] [ERR] switch_ivr_async.c:997 Error opening c:/FreeSWITCH/reco/rec_c1e3df00-dda2-4320-8900-4853a216f19b.wav After that the recorded call gets hung up. Is there any way to continue the call even if recording fails? -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/How-to-continue-call-even-if-… Sent from the freeswitch-users mailing list archive at Nabble.com.

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