November 2009 - FreeSWITCH-users - FreeSWITCH Mailing Lists

SIP Overlap support?
by Dennis 06 Apr '10

06 Apr '10

hi there, i would like to ask, if fs has support for something like "SIP Overlap"? instead of receiving the phonenumber from our carrier in a block, we want to receive the phonenumber digit-by-digit and we want to tell fs when the number is complete. our carrier could send us the phonenumber digit-by-digit, but what about the fs-side? thanks and kind regards dennis

11 29

zrtp endpoints have different sas through fs 1.0.4
by Harondel J. Sibble 19 Jan '10

19 Jan '10

I've got 1.0.4 running with zrtp on ubuntu 9.0.4. I have 3 zrtp capable endpoints: an xp desktop running ekiga with the 0.92 build 218 zfone client, 2 cell phones running ver 2.0.5 of the Tivi softphone: a nokia e61i (symbian s60) and an O2 Xda Flame (windows mobile 5). All 3 endpoints are registered with FS using the default extensions of 1000- 1003 With global_setvar zrtp_secure_media=true the zrtp negotiation between end points happens but the SAS never matches,below is console output for a call between 2 of the endpoints 2009-08-23 14:10:17.643073 [NOTICE] mod_sofia.c:1509 Pre-Answer sofia/internal/1003(a)10.12.13.45! 2009-08-23 14:10:21.257568 [NOTICE] sofia.c:3794 Channel [sofia/internal/sip:1000@10.12.13.166:5062] has been answered 2009-08-23 14:10:21.275521 [NOTICE] switch_ivr_originate.c:2015 Channel [sofia/internal/1003(a)10.12.13.45] has been answered 2009-08-23 14:10:22.232053 [WARNING] mod_sofia.c:810 We were told to use ptime 20 but what they meant to say was 80 This issue has so far been identified to happen on the following broken platforms/devices: Linksys/Sipura aka Cisco ShoreTel Sonus/L3 We will try to fix it but some of the devices on this list are so broken who knows what will happen.. 2009-08-23 14:11:34.496118 [NOTICE] sofia.c:322 Hangup sofia/internal/sip:1000@10.12.13.166:5062 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] 2009-08-23 14:11:34.512100 [NOTICE] switch_ivr_bridge.c:1016 Hangup sofia/internal/1003(a)10.12.13.45 [CS_EXECUTE] [NORMAL_CLEARING] 2009-08-23 14:11:34.552158 [NOTICE] switch_core_session.c:1086 Session 16 (sofia/internal/sip:1000@10.12.13.166:5062) Ended 2009-08-23 14:11:34.552158 [NOTICE] switch_core_session.c:1088 Close Channel sofia/internal/sip:1000@10.12.13.166:5062 [CS_DESTROY] 2009-08-23 14:11:34.556441 [NOTICE] switch_core_session.c:1086 Session 15 (sofia/internal/1003(a)10.12.13.45) Ended 2009-08-23 14:11:34.556441 [NOTICE] switch_core_session.c:1088 Close Channel sofia/internal/1003(a)10.12.13.45 [CS_DESTROY] Of note, with the endpoints registered through the Ekiga sip server, the sas DOES match on both ends. With global_setvar zrtp_secure_media=false, the endpoints can't detect a zrtp peer. Reading the list archives hasn't enlightened me. I see this comment from 2008 http://www.nabble.com/Freeswitch-and-Twinkle-and-ZRTP- td18518140.html#a18518343 On Jul 17, 2008, at 4:23 PM, Michael Jerris wrote: > it should in bypass_media or proxy_media modes. in the other modes we > are in the media path and would not know how to handle the encrypted > packets. > > Mike Is this still relevant? Or is there some other setting not covered here http://wiki.freeswitch.org/wiki/ZRTP to make this work properly? I ask firstly about this in the context of a peer 2 peer zrtp communication between the endpoints, then secondly in the case of FS acting as a trusted middleman as in section 2 here http://www.zfoneproject.com/docs/asterisk/man/html/u_guide.html#passthrough Lastly how does one implement the security enrollment as noted above with FS -- Harondel J. Sibble Sibble Computer Consulting Creating Solutions for the small and medium business computer user. help(a)pdscc.com (use pgp keyid 0x3AD5C11D) http://www.pdscc.com (604) 739-3709 (voice)

3 28

Hardware echo cancellation.
by Tim Uckun 13 Jan '10

13 Jan '10

I am about to build a new machine as a VOIP server. I am going to get either a quad core intel or a six core AMD processor with at least eight gigabytes of RAM in it. Given that much horsepower I am wondering if there is any need to purchase hardware with echo cancellation (I am thinking about redfone devices).. I can save some money by not getting the echo cancellation. So is it worth saving that money? Is it always better to have hardware echo cancellation? Is a quad core capable of dealing with echo cancellation needs of an IVR which is going to take lots of simultaneous calls? Thanks.

10 15

ATA that supports TLS/SRTP w FS
by Mark Campbell-Smith 27 Dec '09

27 Dec '09

HI All, Has anyone got some recommendations on which ATA to buy that supports TLS and SRTP? Thanks!

7 21

Faxing Advice
by Joseph L. Casale 23 Dec '09

23 Dec '09

I need to make faxing easy for some very computer illiterate folk. I am using an email service and going to use procmail to print anything incoming automatically but they cant get the hang of scanning to an email app, so I am going to buy a Linksys PAP2T as per the wiki. Since the setup will never receive inbound remote faxes, I just need to direct all fax's sent from the FXS port (that extension) to the email script in the wiki substituting the destination # as the alias portion of the email. So if I create a dialplan that catches the caller_id_number of the FXS port, does the $1 variable exist in the following scenario: <action application="system" data="/opt/freeswitch/scripts/emailfax.sh $1(a)fax.com /tmp/${uuid}.rxfax.tiff"/> as that's how our service requires fax's, the 10 digit # at their domain, fax.com. Is this a plausible setup? Lastly, I see in the interop list that Audiocodes Mediapack 114 is supported, but the 202 is not listed, is that simply because its new or is it known to not work? Given that its the same price as the Linksys, I would rather get it. Thanks! jlc

5 12

WARNING On Inbound Call Question
by Jerry Richards 23 Dec '09

23 Dec '09

I have my Freeswitch server with an installed Sangoma A101D card. Most everything works okay, however, when I get an inbound call from the PSTN, I see the following warning show up in the log. Additionally, the caller (on the PSTN) does not hear ringback, and if the call is not answered within about 12 seconds, the call ends (so it doesn't go to voice mail). If I make a call from one internal phone to another, then it will go to voice mail after 30 seconds. Here are the two warnings: [WARNING] ss7_boost_client.c:218 TX EVENT (N): CALL_START_ACK:(81) [w1g1] Rc=0 CSid=0 Seq=11 [WARNING] mod_openzap.c:761 VETO Changing state on 1:1 from PROGRESS to PROGRESS_MEDIA Here is the log of the warning upon an inbound call: freeswitch(a)TeoProxy.greyhawk.tonecommander.com> freeswitch(a)TeoProxy.greyhawk.tonecommander.com> freeswitch(a)TeoProxy.greyhawk.tonecommander.com> freeswitch(a)TeoProxy.greyhawk.tonecommander.com> freeswitch(a)TeoProxy.greyhawk.tonecommander.com> freeswitch(a)TeoProxy.greyhawk.tonecommander.com> freeswitch(a)TeoProxy.greyhawk.tonecommander.com> 2009-11-02 09:06:01.664835 [WARNING] ozmod_ss7_boost.c:1141 RX EVENT: CALL_START:(80) [w1g1] CSid=0 Seq=12 Cn=[N/A] Cd=[5384] Ci=[4253813176] 2009-11-02 09:06:01.665824 [DEBUG] ozmod_ss7_boost.c:655 Changing state on 1:1 from DOWN to RING 2009-11-02 09:06:01.665824 [DEBUG] ozmod_ss7_boost.c:841 1:1 STATE [RING] 2009-11-02 09:06:01.665824 [DEBUG] mod_openzap.c:1481 got clear channel sig [START] 2009-11-02 09:06:01.665824 [DEBUG] mod_openzap.c:344 Set codec PCMU 20ms 2009-11-02 09:06:01.665824 [DEBUG] mod_openzap.c:1184 Connect inbound channel OpenZAP/1:1/5384 2009-11-02 09:06:01.665824 [NOTICE] switch_channel.c:602 New Channel OpenZAP/1:1/5384 [b678f311-ab74-4cc1-afac-b83d89a53132] 2009-11-02 09:06:01.665824 [DEBUG] mod_openzap.c:1192 (OpenZAP/1:1/5384) State Change CS_NEW -> CS_INIT 2009-11-02 09:06:01.665824 [DEBUG] switch_core_session.c:932 Send signal OpenZAP/1:1/5384 [BREAK] 2009-11-02 09:06:01.665824 [DEBUG] switch_core_state_machine.c:398 (OpenZAP/1:1/5384) Running State Change CS_INIT 2009-11-02 09:06:01.665824 [DEBUG] switch_core_state_machine.c:481 (OpenZAP/1:1/5384) State INIT 2009-11-02 09:06:01.665824 [DEBUG] mod_openzap.c:368 (OpenZAP/1:1/5384) State Change CS_INIT -> CS_ROUTING 2009-11-02 09:06:01.665824 [DEBUG] switch_core_session.c:932 Send signal OpenZAP/1:1/5384 [BREAK] 2009-11-02 09:06:01.665824 [DEBUG] switch_core_state_machine.c:481 (OpenZAP/1:1/5384) State INIT going to sleep 2009-11-02 09:06:01.665824 [DEBUG] switch_core_state_machine.c:398 (OpenZAP/1:1/5384) Running State Change CS_ROUTING 2009-11-02 09:06:01.665824 [DEBUG] switch_core_state_machine.c:484 (OpenZAP/1:1/5384) State ROUTING 2009-11-02 09:06:01.665824 [DEBUG] mod_openzap.c:391 OpenZAP/1:1/5384 CHANNEL ROUTING 2009-11-02 09:06:01.665824 [DEBUG] switch_core_state_machine.c:78 OpenZAP/1:1/5384 Standard ROUTING 2009-11-02 09:06:01.665824 [INFO] mod_dialplan_xml.c:315 Processing 4253813176->5384 in context default Dialplan: OpenZAP/1:1/5384 parsing [default->unloop] continue=false Dialplan: OpenZAP/1:1/5384 Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false Dialplan: OpenZAP/1:1/5384 parsing [default->tod_example] continue=true Dialplan: OpenZAP/1:1/5384 Absolute Condition [tod_example] Dialplan: OpenZAP/1:1/5384 Action set(open=true) Dialplan: OpenZAP/1:1/5384 parsing [default->SangomaPRI] continue=false Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [SangomaPRI] destination_number(5384) =~ /^9(\d+)$/ break=on-false Dialplan: OpenZAP/1:1/5384 parsing [default->global-intercept] continue=false Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [global-intercept] destination_number(5384) =~ /^(5380)$/ break=on-false Dialplan: OpenZAP/1:1/5384 parsing [default->group-intercept] continue=false Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [group-intercept] destination_number(5384) =~ /^\*8$/ break=on-false Dialplan: OpenZAP/1:1/5384 parsing [default->intercept-ext] continue=false Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [intercept-ext] destination_number(5384) =~ /^\*\*(\d+)$/ break=on-false Dialplan: OpenZAP/1:1/5384 parsing [default->redial] continue=false Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [redial] destination_number(5384) =~ /^870$/ break=on-false Dialplan: OpenZAP/1:1/5384 parsing [default->global] continue=true Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [global] ${call_debug}(false) =~ /^true$/ break=never Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [global] ${sip_has_crypto}() =~ /^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$/ break=never Dialplan: OpenZAP/1:1/5384 Absolute Condition [global] Dialplan: OpenZAP/1:1/5384 Action hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid}) Dialplan: OpenZAP/1:1/5384 Action hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_numbe r}) Dialplan: OpenZAP/1:1/5384 Action hash(insert/${domain_name}-last_dial/global/${uuid}) Dialplan: OpenZAP/1:1/5384 parsing [default->snom-demo-2] continue=false Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [snom-demo-2] destination_number(5384) =~ /^9001$/ break=on-false Dialplan: OpenZAP/1:1/5384 parsing [default->snom-demo-1] continue=false Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [snom-demo-1] destination_number(5384) =~ /^9000$/ break=on-false Dialplan: OpenZAP/1:1/5384 parsing [default->eavesdrop] continue=false Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [eavesdrop] destination_number(5384) =~ /^88(.*)$|^\*0(.*)$/ break=on-false Dialplan: OpenZAP/1:1/5384 parsing [default->eavesdrop] continue=false Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [eavesdrop] destination_number(5384) =~ /^779$/ break=on-false Dialplan: OpenZAP/1:1/5384 parsing [default->call_return] continue=false Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [call_return] destination_number(5384) =~ /^\*69$|^869$|^lcr$/ break=on-false Dialplan: OpenZAP/1:1/5384 parsing [default->del-group] continue=false Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [del-group] destination_number(5384) =~ /^80(\d{2})$/ break=on-false Dialplan: OpenZAP/1:1/5384 parsing [default->add-group] continue=false Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [add-group] destination_number(5384) =~ /^81(\d{2})$/ break=on-false Dialplan: OpenZAP/1:1/5384 parsing [default->call-group-simo] continue=false Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [call-group-simo] destination_number(5384) =~ /^82(\d{2})$/ break=on-false Dialplan: OpenZAP/1:1/5384 parsing [default->call-group-order] continue=false Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [call-group-order] destination_number(5384) =~ /^83(\d{2})$/ break=on-false Dialplan: OpenZAP/1:1/5384 parsing [default->extension-intercom] continue=false Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [extension-intercom] destination_number(5384) =~ /^8(5[34][8901][0-9])$/ break=on-false Dialplan: OpenZAP/1:1/5384 parsing [default->Local_Extension] continue=false Dialplan: OpenZAP/1:1/5384 Regex (PASS) [Local_Extension] destination_number(5384) =~ /^(5[34][8901][0-9])$/ break=on-false Dialplan: OpenZAP/1:1/5384 Action set(dialed_extension=5384) Dialplan: OpenZAP/1:1/5384 Action export(dialed_extension=5384) Dialplan: OpenZAP/1:1/5384 Action bind_meta_app(1 b s execute_extension::dx XML features) Dialplan: OpenZAP/1:1/5384 Action bind_meta_app(2 b s record_session::/usr/local/freeswitch/recordings/${caller_id_number}.${strft ime(%Y-%m-%d-%H-%M-%S)}.wav) Dialplan: OpenZAP/1:1/5384 Action bind_meta_app(3 b s execute_extension::cf XML features) Dialplan: OpenZAP/1:1/5384 Action set(ringback=${us-ring}) Dialplan: OpenZAP/1:1/5384 Action set(transfer_ringback=local_stream://moh) Dialplan: OpenZAP/1:1/5384 Action set(call_timeout=30) Dialplan: OpenZAP/1:1/5384 Action set(hangup_after_bridge=true) Dialplan: OpenZAP/1:1/5384 Action set(continue_on_fail=true) Dialplan: OpenZAP/1:1/5384 Action hash(insert/${domain_name}-call_return/${dialed_extension}/${caller_id_numbe r}) Dialplan: OpenZAP/1:1/5384 Action hash(insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}) Dialplan: OpenZAP/1:1/5384 Action set(called_party_callgroup=${user_data(${dialed_extension}@${domain_name} var callgroup)}) Dialplan: OpenZAP/1:1/5384 Action hash(insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}) Dialplan: OpenZAP/1:1/5384 Action bridge(user/${dialed_extension}@${domain_name}) Dialplan: OpenZAP/1:1/5384 Action answer() Dialplan: OpenZAP/1:1/5384 Action sleep(1000) Dialplan: OpenZAP/1:1/5384 Action voicemail(default ${domain_name} ${dialed_extension}) 2009-11-02 09:06:01.666685 [DEBUG] switch_core_state_machine.c:114 (OpenZAP/1:1/5384) State Change CS_ROUTING -> CS_EXECUTE 2009-11-02 09:06:01.666685 [DEBUG] switch_core_session.c:932 Send signal OpenZAP/1:1/5384 [BREAK] 2009-11-02 09:06:01.666685 [DEBUG] switch_core_state_machine.c:484 (OpenZAP/1:1/5384) State ROUTING going to sleep 2009-11-02 09:06:01.666685 [DEBUG] switch_core_state_machine.c:398 (OpenZAP/1:1/5384) Running State Change CS_EXECUTE 2009-11-02 09:06:01.666685 [DEBUG] switch_core_state_machine.c:491 (OpenZAP/1:1/5384) State EXECUTE 2009-11-02 09:06:01.666685 [DEBUG] mod_openzap.c:408 OpenZAP/1:1/5384 CHANNEL EXECUTE 2009-11-02 09:06:01.666685 [DEBUG] switch_core_state_machine.c:151 OpenZAP/1:1/5384 Standard EXECUTE EXECUTE OpenZAP/1:1/5384 set(open=true) 2009-11-02 09:06:01.666685 [DEBUG] mod_dptools.c:748 OpenZAP/1:1/5384 SET [open]=[true] EXECUTE OpenZAP/1:1/5384 hash(insert/192.168.72.141-spymap/4253813176/b678f311-ab74-4cc1-afac-b83d89a 53132) EXECUTE OpenZAP/1:1/5384 hash(insert/192.168.72.141-last_dial/4253813176/5384) EXECUTE OpenZAP/1:1/5384 hash(insert/192.168.72.141-last_dial/global/b678f311-ab74-4cc1-afac-b83d89a5 3132) EXECUTE OpenZAP/1:1/5384 set(dialed_extension=5384) 2009-11-02 09:06:01.667682 [DEBUG] mod_dptools.c:748 OpenZAP/1:1/5384 SET [dialed_extension]=[5384] EXECUTE OpenZAP/1:1/5384 export(dialed_extension=5384) 2009-11-02 09:06:01.667682 [DEBUG] mod_dptools.c:886 EXPORT [dialed_extension]=[5384] EXECUTE OpenZAP/1:1/5384 bind_meta_app(1 b s execute_extension::dx XML features) 2009-11-02 09:06:01.667682 [INFO] switch_ivr_async.c:1795 Bound B-Leg: 1 execute_extension::dx XML features EXECUTE OpenZAP/1:1/5384 bind_meta_app(2 b s record_session::/usr/local/freeswitch/recordings/4253813176.2009-11-02-09-06 -01.wav) 2009-11-02 09:06:01.668708 [INFO] switch_ivr_async.c:1795 Bound B-Leg: 2 record_session::/usr/local/freeswitch/recordings/4253813176.2009-11-02-09-06 -01.wav EXECUTE OpenZAP/1:1/5384 bind_meta_app(3 b s execute_extension::cf XML features) 2009-11-02 09:06:01.668708 [INFO] switch_ivr_async.c:1795 Bound B-Leg: 3 execute_extension::cf XML features EXECUTE OpenZAP/1:1/5384 set(ringback=%(2000,4000,440.0,480.0)) 2009-11-02 09:06:01.668708 [DEBUG] mod_dptools.c:748 OpenZAP/1:1/5384 SET [ringback]=[%(2000,4000,440.0,480.0)] EXECUTE OpenZAP/1:1/5384 set(transfer_ringback=local_stream://moh) 2009-11-02 09:06:01.668708 [DEBUG] mod_dptools.c:748 OpenZAP/1:1/5384 SET [transfer_ringback]=[local_stream://moh] EXECUTE OpenZAP/1:1/5384 set(call_timeout=30) 2009-11-02 09:06:01.668708 [DEBUG] mod_dptools.c:748 OpenZAP/1:1/5384 SET [call_timeout]=[30] EXECUTE OpenZAP/1:1/5384 set(hangup_after_bridge=true) 2009-11-02 09:06:01.669681 [DEBUG] mod_dptools.c:748 OpenZAP/1:1/5384 SET [hangup_after_bridge]=[true] EXECUTE OpenZAP/1:1/5384 set(continue_on_fail=true) 2009-11-02 09:06:01.669681 [DEBUG] mod_dptools.c:748 OpenZAP/1:1/5384 SET [continue_on_fail]=[true] EXECUTE OpenZAP/1:1/5384 hash(insert/192.168.72.141-call_return/5384/4253813176) EXECUTE OpenZAP/1:1/5384 hash(insert/192.168.72.141-last_dial_ext/5384/b678f311-ab74-4cc1-afac-b83d89 a53132) EXECUTE OpenZAP/1:1/5384 set(called_party_callgroup=techsupport) 2009-11-02 09:06:01.670679 [DEBUG] mod_dptools.c:748 OpenZAP/1:1/5384 SET [called_party_callgroup]=[techsupport] EXECUTE OpenZAP/1:1/5384 hash(insert/192.168.72.141-last_dial/techsupport/b678f311-ab74-4cc1-afac-b83 d89a53132) EXECUTE OpenZAP/1:1/5384 bridge(user/5384(a)192.168.72.141) 2009-11-02 09:06:01.671683 [DEBUG] switch_ivr_originate.c:1027 variable string 0 = [presence_id=5384(a)192.168.72.141] 2009-11-02 09:06:01.671683 [NOTICE] switch_channel.c:602 New Channel sofia/internal/sip:5384@192.168.72.163:5060 [9e7b8fae-6194-430c-951b-948ebd2c2a3b] 2009-11-02 09:06:01.671683 [DEBUG] mod_sofia.c:2811 (sofia/internal/sip:5384@192.168.72.163:5060) State Change CS_NEW -> CS_INIT 2009-11-02 09:06:01.672688 [DEBUG] switch_core_session.c:932 Send signal sofia/internal/sip:5384@192.168.72.163:5060 [BREAK] 2009-11-02 09:06:01.672688 [DEBUG] switch_core_state_machine.c:398 (sofia/internal/sip:5384@192.168.72.163:5060) Running State Change CS_INIT 2009-11-02 09:06:01.672688 [DEBUG] switch_core_state_machine.c:481 (sofia/internal/sip:5384@192.168.72.163:5060) State INIT 2009-11-02 09:06:01.672688 [DEBUG] mod_sofia.c:83 sofia/internal/sip:5384@192.168.72.163:5060 SOFIA INIT 2009-11-02 09:06:01.672688 [DEBUG] mod_sofia.c:111 (sofia/internal/sip:5384@192.168.72.163:5060) State Change CS_INIT -> CS_ROUTING 2009-11-02 09:06:01.672688 [DEBUG] switch_core_session.c:932 Send signal sofia/internal/sip:5384@192.168.72.163:5060 [BREAK] 2009-11-02 09:06:01.672688 [DEBUG] switch_core_state_machine.c:481 (sofia/internal/sip:5384@192.168.72.163:5060) State INIT going to sleep 2009-11-02 09:06:01.672688 [DEBUG] switch_core_state_machine.c:398 (sofia/internal/sip:5384@192.168.72.163:5060) Running State Change CS_ROUTING 2009-11-02 09:06:01.672688 [DEBUG] switch_core_state_machine.c:484 (sofia/internal/sip:5384@192.168.72.163:5060) State ROUTING 2009-11-02 09:06:01.672688 [DEBUG] mod_sofia.c:130 sofia/internal/sip:5384@192.168.72.163:5060 SOFIA ROUTING 2009-11-02 09:06:01.672688 [DEBUG] switch_ivr_originate.c:63 (sofia/internal/sip:5384@192.168.72.163:5060) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2009-11-02 09:06:01.672688 [DEBUG] switch_core_session.c:932 Send signal sofia/internal/sip:5384@192.168.72.163:5060 [BREAK] 2009-11-02 09:06:01.672688 [DEBUG] switch_core_state_machine.c:484 (sofia/internal/sip:5384@192.168.72.163:5060) State ROUTING going to sleep 2009-11-02 09:06:01.672688 [DEBUG] switch_core_state_machine.c:398 (sofia/internal/sip:5384@192.168.72.163:5060) Running State Change CS_CONSUME_MEDIA 2009-11-02 09:06:01.672688 [DEBUG] sofia.c:3289 Channel sofia/internal/sip:5384@192.168.72.163:5060 entering state [calling][0] 2009-11-02 09:06:01.672688 [DEBUG] switch_core_state_machine.c:503 (sofia/internal/sip:5384@192.168.72.163:5060) State CONSUME_MEDIA 2009-11-02 09:06:01.672688 [DEBUG] switch_ivr_originate.c:1701 OpenZAP/1:1/5384 receive message [PROGRESS] 2009-11-02 09:06:01.673742 [DEBUG] mod_openzap.c:759 Changing state on 1:1 from RING to PROGRESS 2009-11-02 09:06:01.674787 [DEBUG] ozmod_ss7_boost.c:841 1:1 STATE [PROGRESS] 2009-11-02 09:06:01.675844 [WARNING] ss7_boost_client.c:218 TX EVENT (N): CALL_START_ACK:(81) [w1g1] Rc=0 CSid=0 Seq=11 2009-11-02 09:06:01.684776 [WARNING] mod_openzap.c:761 VETO Changing state on 1:1 from PROGRESS to PROGRESS_MEDIA 2009-11-02 09:06:01.684776 [DEBUG] switch_core_session.c:630 Send signal OpenZAP/1:1/5384 [BREAK] 2009-11-02 09:06:01.684776 [NOTICE] switch_ivr_originate.c:1701 Pre-Answer OpenZAP/1:1/5384! 2009-11-02 09:06:01.684776 [DEBUG] switch_ivr_originate.c:1718 Raw Codec Activation Success L16@8000hz 1 channel 20ms 2009-11-02 09:06:01.684776 [DEBUG] switch_ivr_originate.c:1777 Play Ringback Tone [%(2000,4000,440.0,480.0)] 2009-11-02 09:06:01.693835 [DEBUG] sofia.c:3289 Channel sofia/internal/sip:5384@192.168.72.163:5060 entering state [proceeding][180] 2009-11-02 09:06:01.693835 [NOTICE] sofia.c:3353 Ring-Ready sofia/internal/sip:5384@192.168.72.163:5060! 2009-11-02 09:06:01.705777 [DEBUG] switch_core_io.c:649 OpenZAP/1:1/5384 receive message [TRANSCODING_NECESSARY] freeswitch(a)TeoProxy.greyhawk.tonecommander.com> freeswitch(a)TeoProxy.greyhawk.tonecommander.com> freeswitch(a)TeoProxy.greyhawk.tonecommander.com> freeswitch(a)TeoProxy.greyhawk.tonecommander.com> freeswitch(a)TeoProxy.greyhawk.tonecommander.com> freeswitch(a)TeoProxy.greyhawk.tonecommander.com> freeswitch(a)TeoProxy.greyhawk.tonecommander.com> freeswitch(a)TeoProxy.greyhawk.tonecommander.com> freeswitch(a)TeoProxy.greyhawk.tonecommander.com> freeswitch(a)TeoProxy.greyhawk.tonecommander.com> freeswitch(a)TeoProxy.greyhawk.tonecommander.com> Best Regards, Jerry

4 4

How can I join two freeswitch on two servers?
by Edmar Cruz 18 Dec '09

18 Dec '09

I like to connect two freeswitch, call each other, communicate and vice versa. Can you give me an example for that? Thanks -- View this message in context: http://www.nabble.com/How-can-I-join-two-freeswitch-on-two-servers--tp24045… Sent from the Freeswitch-users mailing list archive at Nabble.com.

6 20

Polycom 501 conferencing with FreeSwitch
by Yehavi Bourvine 17 Dec '09

17 Dec '09

Hello, I am trying to set a Polycom 501 phone to do conferencing via the conference room on Freeswitch rather than on the phone (as on the phone it is limited to 3 participants only). Anyone had success with it? I have on the Freeswitch an extension named Conf.* which activates the conference application (it works with other brands). On the Polycom I tried to define voIpProt.SIP.*conference*.address=sip:Conf0000@freeswitch-server. The phone continues to create the conference locally and add the above Conf0000 to it, without REFERing the parties to it. The first phone which called is left on hold... Anyone managed to make this feature work? We use firmware 3.1.3. Thanks! __Yehavi:

3 5

mod_shout.so: undefined symbol: ogg_sync_wrote
by Sean Ferguson 17 Dec '09

17 Dec '09

FreeSWITCH seems to be unable to read MP3 files, citing that it's an unknown format. Looking through the log, I found this during startup: 2009-11-07 02:43:45.749328 [CRIT] switch_loadable_module.c:871 Error Loading module /usr/local/freeswitch/mod/mod_shout.so **/usr/local/freeswitch/mod/mod_shout.so: undefined symbol: ogg_sync_wrote** There don't seem to be any compile-time errors, yet I can't seem to eliminate this issue. Any help would be appreciated.

4 6

Problem while playing more than 10 voice files using playback
by Thangappan.M 12 Dec '09

12 Dec '09

Dear all, I am in the process of implementing IVR using event outbound socket (async mode). I have implemented using Perl language. I did the following steps: => Set the playback_delimiter variable => Set the playback_sleep_val variable => Set the event lock as true => Set the freeswitch ( my own) variable as zero => Wait in the loop until the variable is been set as zero => Playback the voice files ( Here I combined the voice files with the delimiter value if more than one voice files are there) => Set the freeswitch(my own) variable as true ( This is used to identify whether the voice files are played successfully). => Wait in the loop until the variable is been set as one. => Set the Event lock as false => Trying to get the DTMF digits ( Have a assurance that all the voice files are played). The problem is, The above steps are working fine when the voice file count is lesser than or equal to 10. After the voice files are played only the variable(my own freeswitch) is set. Based on the variable I am doing further things. But when I tried to give the voice files count of more than 10 the variable has been set while starting to play back the first voice file itself . Because of this I am not able to proceed further. *DID I MAKE ANY MISTAKE IN THE ABOVE STEPS?* *NOTE*: I also referred mod_file_string documentation. In that they specified 128 files can be used to play back the voice files using playback_delimiter option. Please help me................? Thanks in advance. -- Regards, Thangappan.M

3 9